ASTERWEB Blog

Rilasciato Asterisk 13.7.0-rc2

Il giorno 18 dicembre 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.7.0-rc2.

Dal post originale:

Bug

[ASTERISK-25601] - json: Audit reference usage and thread safety
[ASTERISK-25625] - res_sorcery_memory_cache: Add full backend caching

Per la lista completa, questo il link al ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.0-rc2

Rilasciato Asterisk 13.7.0-rc1

Il giorno 15 dicembre 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.7.0-rc1.

Dal post originale:

Bug

[ASTERISK-7803] - [patch] Update the maximum packetization values in frame.c
[ASTERISK-24106] - WebSockets Automatically decides what driver it will use
[ASTERISK-24146] - [patch]No audio on WebRtc caller side when answer waiting time is more than ~7sec
[ASTERISK-24543] - Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs
[ASTERISK-24779] - Passthrough OPUS codec not working with chan_pjsip
[ASTERISK-24958] - Forwarding loop detection inhibits certain desirable scenarios
[ASTERISK-25135] - [patch]RTP Timeout hangup cause code missing
[ASTERISK-25160] - [patch] Opus Codec: SIP/SDP line fmtp missing when called internally
[ASTERISK-25165] - Testsuite - Sorcery memory cache leaks
[ASTERISK-25364] - [patch]Issue a TCP connection(kernel) and thread of asterisk is not released
[ASTERISK-25373] - add documentation for CALLERID(pres) and also the CONNECTEDLINE and REDIRECTING variants
[ASTERISK-25391] - AMI GetConfigJSON returns invalid JSON
[ASTERISK-25400] - Hints broken when "CustomPresence" doesn't exist in AstDB
[ASTERISK-25404] - segfault/crash in chan_pjsip_hangup ... at chan_pjsip.c
[ASTERISK-25434] - Compiler flags not reported in 'core show settings' despite usage during compilation
[ASTERISK-25435] - Asterisk periodically hangs. UDP Recv-Q greatly exceeds zero.
[ASTERISK-25438] - res_rtp_asterisk: ICE role message even when ICE is not enabled
[ASTERISK-25441] - Deadlock in res_sorcery_memory_cache.
[ASTERISK-25443] - [patch]IPv6 - Potential issue in via header parsing
[ASTERISK-25449] - main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny
[ASTERISK-25451] - Broken video - erased rtp marker bit
[ASTERISK-25455] - Deadlock of PJSIP realtime over res_config_pgsql
[ASTERISK-25461] - Nested dialplan #includes don't work as expected.
[ASTERISK-25476] - chan_sip loses registrations after a while
[ASTERISK-25484] - [patch] autoframing=yes has no effect
[ASTERISK-25485] - res_pjsip_outbound_registration: registration stops due to 400 response
[ASTERISK-25486] - res_pjsip: Fix deadlock when validating URIs
[ASTERISK-25494] - build: GCC 5.1.x catches some new const, array bounds and missing paren issues
[ASTERISK-25498] - Asterisk crashes when negotiating g729 without that module installed
[ASTERISK-25505] - res_pjsip_pubsub: Crash on off-nominal when UAS dialog can't be created
[ASTERISK-25513] - Crash: malloc failed with high load of subscriptions.
[ASTERISK-25522] - ARI: Crash when creating channel via ARI originate with requesting channel
[ASTERISK-25527] - Quirky xmldoc description wrapping
[ASTERISK-25533] - [patch] buffer for ast_format_cap_get_names only 64 bytes
[ASTERISK-25535] - [patch] format creation on module load instead of cache
[ASTERISK-25537] - [patch] format-attribute module: RFC or internal defaults?
[ASTERISK-25545] - [patch] translation module gets cached not joint format
[ASTERISK-25546] - threadpool: Race condition between idle timeout and activation
[ASTERISK-25552] - hashtab: Improve NULL tolerance
[ASTERISK-25561] - app_queue.c line 6503 (try_calling): mutex 'qe->chan' freed more times than we've locked!
[ASTERISK-25569] - app_meetme: Audio quality issues
[ASTERISK-25573] - [patch] H.264 format attribute module: resets whole SDP
[ASTERISK-25575] - res_pjsip: Dynamic outbound registrations created via ARI are not loaded into memory on Asterisk start/restart
[ASTERISK-25582] - Testsuite: Reactor timeout error in tests/fax/pjsip/directmedia_reinvite_t38
[ASTERISK-25583] - [patch] format-attribute module: RFC 7587 (Opus Codec)
[ASTERISK-25584] - [patch] format-attribute module: VP8 missing
[ASTERISK-25585] - [patch]rasterisk never hits most of main(), but it's assumed to
[ASTERISK-25590] - CLI Usage info for 'pjsip send notify' references incorrect config
[ASTERISK-25593] - fastagi: record file closed after sending result
[ASTERISK-25595] - Unescaped : in messge sent to statsd
[ASTERISK-25598] - res_pjsip: Contact status messages are printing a hash instead of the uri
[ASTERISK-25599] - [patch] SLIN Resampling Codec only 80 msec
[ASTERISK-25600] - bridging: Inconsistency in BRIDGEPEER
[ASTERISK-25608] - res_pjsip/contacts/statsd: Lifecycle events aren't consistent
[ASTERISK-25609] - [patch]Asterisk may crash when calling ast_channel_get_t38_state(c)
[ASTERISK-25610] - Asterisk crash during "sip reload"
[ASTERISK-25615] - res_pjsip: Setting transport async_operations > 1 causes segfault on tls transports
[ASTERISK-25616] - Warning with a Codec Module which supports PLC with FEC
[ASTERISK-25619] - res_chan_stats not sending the correct information to StatsD

Improvement

[ASTERISK-24718] - [patch]Add inital support of "sanitize" to configure
[ASTERISK-25477] - pjsip show "command" like [criteria]
[ASTERISK-25518] - taskprocessor: Add high water mark
[ASTERISK-25571] - PJSIP: Add StatsD stats for some common PJSIP objects
[ASTERISK-25572] - Endpoints: Add StatsD stats for Asterisk endpoints
[ASTERISK-25618] - res_pjsip: Check for readability of TLS files at startup

New Feature

[ASTERISK-24922] - ARI: Add the ability to intercept hold and raise an event
[ASTERISK-25419] - Dialplan Application for Integration of StatsD
[ASTERISK-25549] - Confbridge: Add participant timeout option

Per la lista completa, questo il link al ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.0-rc1

Rilasciato Asterisk 11.21.0-rc1

Il giorno 15 dicembre 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 11.21.0-rc1.

Dal post originale:

Bug

[ASTERISK-7803] - [patch] Update the maximum packetization values in frame.c
[ASTERISK-24146] - [patch]No audio on WebRtc caller side when answer waiting time is more than ~7sec
[ASTERISK-25135] - [patch]RTP Timeout hangup cause code missing
[ASTERISK-25364] - [patch]Issue a TCP connection(kernel) and thread of asterisk is not released
[ASTERISK-25373] - add documentation for CALLERID(pres) and also the CONNECTEDLINE and REDIRECTING variants
[ASTERISK-25391] - AMI GetConfigJSON returns invalid JSON
[ASTERISK-25400] - Hints broken when "CustomPresence" doesn't exist in AstDB
[ASTERISK-25434] - Compiler flags not reported in 'core show settings' despite usage during compilation
[ASTERISK-25438] - res_rtp_asterisk: ICE role message even when ICE is not enabled
[ASTERISK-25443] - [patch]IPv6 - Potential issue in via header parsing
[ASTERISK-25449] - main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny
[ASTERISK-25455] - Deadlock of PJSIP realtime over res_config_pgsql
[ASTERISK-25461] - Nested dialplan #includes don't work as expected.
[ASTERISK-25476] - chan_sip loses registrations after a while
[ASTERISK-25494] - build: GCC 5.1.x catches some new const, array bounds and missing paren issues
[ASTERISK-25498] - Asterisk crashes when negotiating g729 without that module installed
[ASTERISK-25527] - Quirky xmldoc description wrapping
[ASTERISK-25537] - [patch] format-attribute module: RFC or internal defaults?
[ASTERISK-25552] - hashtab: Improve NULL tolerance
[ASTERISK-25569] - app_meetme: Audio quality issues
[ASTERISK-25585] - [patch]rasterisk never hits most of main(), but it's assumed to
[ASTERISK-25593] - fastagi: record file closed after sending result
[ASTERISK-25599] - [patch] SLIN Resampling Codec only 80 msec
[ASTERISK-25609] - [patch]Asterisk may crash when calling ast_channel_get_t38_state(c)
[ASTERISK-25610] - Asterisk crash during "sip reload"
[ASTERISK-25616] - Warning with a Codec Module which supports PLC with FEC

Improvement

[ASTERISK-24718] - [patch]Add inital support of "sanitize" to configure

Per la lista completa, questo il link al ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0-rc1

Bridges, T.38, and other good times

Matt Jordan, il 06 dicembre 2915, ha pubblicato un "lunghissimo" post con una accurata "discussione" circa il (non)funzionamento di fax/pjsip/directmedia_reinvite_t38

Questo il link del post:
https://www.mail-archive.com/asterisk-dev@lists.digium.com/msg42520.html

Feedback sui ​​miglioramenti ODBC in Asterisk by Mark Michelson

Mark Michelson, il 30 novembre 2915, ha pubblicato una richiesta di un feedback sui ​​miglioramenti ODBC in Asterisk ed ha affrontato interessanti tematiche relative all'utilizzo del pjsip rispetto alle commessioni tramite ODBX.

Questo il link del post:
https://www.mail-archive.com/asterisk-dev@lists.digium.com/msg42512.html

Rilasciato Asterisk 14.1.2

Il giorno 10 novembre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 14.1.2.

Dal post originale:

The release of Asterisk 14.1.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Bugs fixed in this release:
-----------------------------------
* ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
calls after 2 minutes - rtptimeout behaving badly - regression
(Reported by Michael Keuter)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.2

Rilasciato Asterisk 13.6.0

Il giorno 09 ottobre 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.6.0.

Dal post originale:

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] - [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

Improvement

[ASTERISK-24870] - ARI: Subscriptions to bridges generally not super useful
[ASTERISK-25310] - [patch]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

Per la lista completa, questo il link al ChangeLog:

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

Rilasciato Asterisk 11.20.0

Il giorno 09 ottobre 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 11.20.0.

Dal post originale:

Bug

[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-25299] - RTP port leaks with incoming OOH323 calls
[ASTERISK-25312] - res_http_websocket: Terminate connection on fatal cases
[ASTERISK-25315] - DAHDI channels send shortened duration DTMF tones.
[ASTERISK-25320] - chan_sip.c: sip_report_security_event searches for wrong or non existent peer on invite
[ASTERISK-25346] - chan_sip: Overwriting answered elsewhere hangup cause on call pickup
[ASTERISK-25353] - [patch] Transcoding while different in Frame size = Frames lost
[ASTERISK-25391] - AMI GetConfigJSON returns invalid JSON
[ASTERISK-25394] - pbx: Incorrect device and presence state when changing hint details
[ASTERISK-25396] - chan_sip: Extremely long callerid name causes invalid SIP
[ASTERISK-25407] - Asterisk fails to log to multiple syslog destinations
[ASTERISK-25410] - app_record: RECORDED_FILE variable not being populated
[ASTERISK-25427] - Callerid change does not always emit NewCallerid AMI event
[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-25310] - [patch]on FreeBSD also pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED

Per la lista completa, questo il link al ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0

Rilasciato Asterisk 11.20.0-rc1

Il giorno 30 settembre 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 11.20.0-rc1.

Dal post originale:

Bug

[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-25299] - RTP port leaks with incoming OOH323 calls
[ASTERISK-25312] - res_http_websocket: Terminate connection on fatal cases
[ASTERISK-25315] - DAHDI channels send shortened duration DTMF tones.
[ASTERISK-25320] - chan_sip.c: sip_report_security_event searches for wrong or non existent peer on invite
[ASTERISK-25346] - chan_sip: Overwriting answered elsewhere hangup cause on call pickup
[ASTERISK-25394] - pbx: Incorrect device and presence state when changing hint details
[ASTERISK-25396] - chan_sip: Extremely long callerid name causes invalid SIP
[ASTERISK-25407] - Asterisk fails to log to multiple syslog destinations
[ASTERISK-25410] - app_record: RECORDED_FILE variable not being populated

Per la lista completa, questo il link al ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0-rc1

RECOVERY TELEFONO VOIP SNOM 710

Girovagando per il web ho trovato questo post che, sperando non serva da mettere in pratica, potrà essere estremamente utile.

Questo il LINK

Lo Staff Asterweb

Digium rilascia Respoke iOS e Android SDKs per WebRTC e Messaging

Il giorno 11 agosto 2015 è stato pubblicato un post da parte di Digium che informa circa il rilascio, da parte della stessa Digium, degli SKD per iOS e Android per l'interfacciamento alla piattaforma Digium’s Respoke.

Questo il link del post:
Digium Releases Respoke iOS and Android SDKs for WebRTC and Messaging

Lo Staff Asterweb

Rilasciato Asterisk 13.5.0

Il giorno 07 agosto 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.5.0.

Dal post originale:
The release of Asterisk 13.5.0 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:

Improvements made in this release:
-----------------------------------
* ASTERISK-25256 - [patch]Post AMI VarSet to empty string events when Asterisk deletes a dialplan variable. (Reported by Richard Mudgett)
* ASTERISK-25067 - Sorcery Caching: Implement a new caching module (Reported by Matt Jordan)
* ASTERISK-25040 - pbx: Improve performance of reloads by making hint destruction more performant (Reported by Matt Jordan)
* ASTERISK-25114 - res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes (Reported by George Joseph)
* ASTERISK-25072 - res_pjsip_outbound_registration: line functionality. Additional check for using the request URI (Reported by Dmitriy Serov)

Bugs fixed in this release:
-----------------------------------
* ASTERISK-25250 - chan_sip - Despite the channel being answered, caller on a call established via Local channel continues to hear ringback (Reported by Etienne Lessard)
* ASTERISK-25253 - confbridge volume options and other volume controls such as func_volume don't work (Reported by Dmitriy Serov)
* ASTERISK-25247 - choppy audio when spying on a g722 channel, chan_sip or chan_pjsip (Reported by hristo)
* ASTERISK-24867 - Docs for 'e' option in ResetCDR say to use CDR_PROP instead, CDR_PROP docs are unclear (Reported by Rusty Newton)

... e tanto altro.

Questo il changelog per vedere l'elenco completo http://lists.digium.com/pipermail/asterisk-announce/2015-August/000607.html

Rilasciato Asterisk 11.19.0

Il giorno 07 agosto 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 11.19.0.

Dal post originale:
The release of Asterisk 11.19.0 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:

Bugs fixed in this release:
-----------------------------------
* ASTERISK-25250 - chan_sip - Despite the channel being answered, caller on a call established via Local channel continues to hear ringback (Reported by Etienne Lessard)
* ASTERISK-25247 - choppy audio when spying on a g722 channel, chan_sip or chan_pjsip (Reported by hristo)
* ASTERISK-24853 - Documentation claims chan_sip outbound registrations support WS or WSS as valid transports (not true) (Reported by PSDK)
* ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span -> force_restart_unavailable_chans in wrong scope (Reported by Patric Marschall)
* ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes (Reported by Rusty Newton)
* ASTERISK-22805 - res_rtp_asterisk: Crash when calling BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP (Reported by Dmitry Burilov)
* ASTERISK-24550 - res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handshake (Reported by Osaulenko Alexander)
* ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by Badalian Vyacheslav)
* ASTERISK-24832 - [patch]DTLS-crashes within openssl (Reported by Stefan Engstr)
* ASTERISK-25127 - DTLS crashes following "Unable to cancel schedule ID" in dtls_srtp_check_pending (Reported by Dade Brandon)
* ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip INVITE early Replace code (Reported by Walter Doekes)
* ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c (Reported by Walter Doekes)
* ASTERISK-25219 - [patch]Source and destination overlap in memcpy in rtp_engine.c (Reported by Walter Doekes)
* ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS (Reported by Walter Doekes)
* ASTERISK-19277 - [patch]endlessly repeating error: "poll failed: Bad file descriptor" (Reported by Barry Chern)
* ASTERISK-25202 - Hints extension state broken between 13.3.2 and 13.4 (Reported by cervajs)
* ASTERISK-25154 - [patch]fromtag may need to be updated after successful call dialog match (Reported by Damian Ivereigh)
* ASTERISK-25139 - Malicious transfer sequence locks up Asterisk (Reported by Gregory Massel)
* ASTERISK-25094 - PBX core: Investigate thread safety issues (Reported by Corey Farrell)
* ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute but asterisk doesn't detect it. (Reported by ibercom)
* ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10} (Reported by Badalian Vyacheslav)
* ASTERISK-25100 - asterisk coredump if host has an IPv6 address that end with ::80 (Reported by Mark Petersen)

Questo il changelog per vedere l'elenco completo http://lists.digium.com/pipermail/asterisk-announce/2015-August/000606.html