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.
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
Corso Asterisk per programmazione Web
Il nuovo corso, in calendario dal 28 al 30 settembre 2015, ha come titolo: "Web Application via Socket Manager".
E' un corso rivolto a tutti coloro che desiderano sviluppare in proprio web applications basate con l'interazione con Astrerisk via socket.
Questo il programma del corso:
- Creazione dell'ambiente di sviluppo lato Server
- Installazione delle librerie necessarie
- Panoramica circa l'utilizzo delle librerie installate
- Creazione delle applicazioni lato server
- Creazione delle applicazioni lato client
- Funzionamento delle librerie per Asterisk
- Gestione degli eventi del Manager di Asterisk
- Comunicazione server/client e viceversa
Per i dettagli del corso: CLICK QUI
Buon lavoro
Lo Staff Asterweb
Aggiornamento sezione tutorials/guide del sito Asterweb
Dal 25 luglio 2015 abbiamo iniziato l'aggiornamento della sezione "Tutorials/Guide" del nostro sito www.asterweb.org
Oltre all'aggiornamento abbiamo iniziato ad inserire nuovi tutorials/guide che, siamo certi, troverete interessanti.
Questa attività di aggiornamento proseguirà settimanalmente, per i seguenti argomenti:
- Asterisk
- FreePBX
- Linux
Buon lavoro a tutti.
Lo Staff Asterweb
Digium abbandona il progetto Asterisk-GUI
Digium ha rimosso il repository del progetto Digium Asterisk-GUI Asterisk.
Il progetto negli ultimi quattro anni non era mai stato aggiornato (correzioni di bue e miglioramenti non erano mai stati fatti).
Ma il motivo principale che ha portato alla chiusura è l'incompatibilità con le ultime versioni di Asterisk.
Malcolm Davenport di Digium ha così commentato:
Asterisk-GUI è stato rimosso dal server dei download Digium e Asterisk. E 'un progetto morto che non è stato più stato aggiornato negli ultimi quattro anni, che non ha ricevuto correzioni di bug e miglioramenti e che non è funzionale con le versioni moderne di Asterisk.
Rilasciato Asterisk 13.4.0
Il giorno 14 giugno 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.4.0.
Dal post originale:
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-24922 - ARI: Add the ability to intercept hold and
raise an event (Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25112 - Logger: Configuration settings are not reset to default during reload. (Reported by Corey Farrell)
* ASTERISK-24944 - main/audiohook.c change prevents G722 call recording (Reported by Ronald Raikes)
* ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2 or more digits (Reported by Makoto Dei)
* ASTERISK-25086 - [patch]PJSIP crashes if endpoint missing in Dial() (Reported by snuffy)
* ASTERISK-25089 - res_pjsip_config_wizard: Variable specified in templates aren't being processed correctly (Reported by George Joseph)
* ASTERISK-25090 - CLI core show channel truncates cdr variables (Reported by snuffy)
* ASTERISK-25085 - [patch]Potential crash after unload of func_periodic_hook or test_message (Reported by Corey Farrell)
* ASTERISK-25083 - Message.c: Message channel becomes saturated with frames leading to spammy log messages (Reported by Jonathan Rose)
* ASTERISK-25082 - Asterisk deletes message after doing a playback of an INBOX message using ast_vm_play when the Old folder is full for that mailbox. (Reported by Jonathan Rose)
* ASTERISK-25041 - [patch]Broken column type checking in res_config_mysql addon (Reported by Alexandre Fournier)
* ASTERISK-21893 - Segfault after call hangup, in ast_channel_hangupcause_set, at channel_internal_api.c (Reported by Alexandr Gordeev)
* ASTERISK-25074 - Regression: Recent clang-related change broke cross compiling of Asterisk (Reported by Sebastian Kemper)
* ASTERISK-25042 - asterisk.conf options override command-line options. (Reported by Corey Farrell)
* ASTERISK-24442 - Outgoing call files don't work properly when set in the future (Reported by tootai)
* ASTERISK-25057 - res_pjsip_pubsub: Crash in send_notify due to invalid root pointer in sub_tree (Reported by Matt Jordan)
* ASTERISK-24938 - ARI Snoop Channel results in excessive escalating CPU usage (Reported by George Ladoff)
* ASTERISK-25034 - chan_dahdi: Some telco switches occasionally ignore ISDN RESTART requests. (Reported by Richard Mudgett)
* ASTERISK-25003 - Asterisk crashes on attended transfer (using feature) (Reported by Artem Volodin)
* ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always contain waiting time (Reported by Etienne Lessard)
* ASTERISK-25027 - Build System: Many ARI modules are missing dependencies. (Reported by Corey Farrell)
* ASTERISK-25061 - pbx_config: Register manager actions with module version of macro. (Reported by Corey Farrell)
* ASTERISK-25025 - Periodic crashes (in ast_channel_snapshot_create at stasis_channels.c) with Certified Asterisk 13. (Reported by Chet Stevens)
* ASTERISK-25053 - Unit test category /main/presence missing trailing slash. (Reported by Corey Farrell)
* ASTERISK-22708 - res_odbc.conf negative_connection_cache option not respected, failover between DSNs doesn't work (Reported by JoshE)
... e tanto altro.
Questo il changelog per vedere l'elenco completo http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.4.0
Rilasciato Asterisk 11.18.0
Il giorno 04 giugno 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 11.18.0.
Dal post originale:
The following are the issues resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25112 - Logger: Configuration settings are not reset to default during reload. (Reported by Corey Farrell)
* ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2 or more digits (Reported by Makoto Dei)
* ASTERISK-24944 - main/audiohook.c change prevents G722 call recording (Reported by Ronald Raikes)
* ASTERISK-25083 - Message.c: Message channel becomes saturated with frames leading to spammy log messages (Reported by Jonathan Rose)
* ASTERISK-25041 - [patch]Broken column type checking in res_config_mysql addon (Reported by Alexandre Fournier)
* ASTERISK-21893 - Segfault after call hangup, in ast_channel_hangupcause_set, at channel_internal_api.c (Reported by Alexandr Gordeev)
* ASTERISK-25074 - Regression: Recent clang-related change broke cross compiling of Asterisk (Reported by Sebastian Kemper)
* ASTERISK-25042 - asterisk.conf options override command-line options. (Reported by Corey Farrell)
* ASTERISK-24442 - Outgoing call files don't work properly when set in the future (Reported by tootai)
* ASTERISK-25034 - chan_dahdi: Some telco switches occasionally ignore ISDN RESTART requests. (Reported by Richard Mudgett)
* ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always contain waiting time (Reported by Etienne Lessard)
* ASTERISK-22708 - res_odbc.conf negative_connection_cache option not respected, failover between DSNs doesn't work (Reported by JoshE)
* ASTERISK-25028 - Build System: Unneeded defines in asterisk/buildopts.h (Reported by Corey Farrell)
* ASTERISK-19608 - Asterisk-1.8.x starts rejecting calls with cause code 44 after some time. (Reported by Denis Alberto Martinez)
* ASTERISK-24976 - cdr_odbc not include new columns added on 1.8 (Reported by Rodrigo Ramirez Norambuena)
... e tanto altro.
Questo il changelog per vedere l'elenco completo http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
Rilasciato CLASS – Modulo beta – Monitoraggio Code in Realtime
Logo Asterweb
Abbiamo rilasciato il modulo Code Monitoraggio in Realtime" in versione beta. Questo uno screenshot:
Il rilascio stabile è previsto per il giorno 20 maggio con anche le statistiche sulle code.
Lo Staff Asterweb
AST-2015-003: TLS Certificate Common name NULL byte exploit
Il giorno 08 aprile 2015, l'Asterisk Security Team ha rilasciato l'annunciato di sicurezza visualizzabile da questo link:
http://lists.digium.com/pipermail/asterisk-announce/2015-April/000600.html
Rilasciato Asterisk 13.3.1
Il giorno 06 aprile 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.3.1.
Dal post originale:
The release of Asterisk 13.3.1 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:
* --- pjsip: resolve compatibility problem with ast_sip_session
(Closes issue ASTERISK-24941. Reported by Matt Jordan)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.1
Rilasciato Asterisk 13.3.0
Il giorno 01 aprile 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.3.0.
Dal post originale:
The release of Asterisk 13.3.0 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-24703 - ARI: Add the ability to "transfer" (redirect) a
channel (Reported by Matt Jordan)
* ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
(Reported by Dwayne Hubbard)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-24616 - Crash in res_format_attr_h264 due to invalid
string copy (Reported by Yura Kocyuba)
* ASTERISK-24748 - res_pjsip: If wizards explicitly configured in
sorcery.conf false ERROR messages may occur (Reported by Joshua
Colp)
* ASTERISK-24769 - res_pjsip_sdp_rtp: Local ICE candidates leaked
(Reported by Matt Jordan)
* ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
res_odbc (Reported by ibercom)
* ASTERISK-24479 - Enable REF_DEBUG for module references
(Reported by Corey Farrell)
* ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
fully disconnect underlying socket, leading to events being
dropped with no additional information (Reported by Matt Jordan)
* ASTERISK-24772 - ODBC error in realtime sippeers when device
unregisters under MariaDB (Reported by Richard Miller)
* ASTERISK-24752 - Crash in bridge_manager_service_req when bridge
is destroyed by ARI during shutdown (Reported by Richard
Mudgett)
* ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported
by Zane Conkle)
* ASTERISK-24015 - app_transfer fails with PJSIP channels
(Reported by Private Name)
* ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk
transfer scenario. (Reported by Mark Michelson)
* ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by
Niklas Larsson)
* ASTERISK-24716 - Improve pjsip log messages for presence
subscription failure (Reported by Rusty Newton)
* ASTERISK-24612 - res_pjsip: No information if a required sorcery
wizard is not loaded (Reported by Joshua Colp)
* ASTERISK-24768 - res_timing_pthread: file descriptor leak
(Reported by Matthias Urlichs)
* ASTERISK-24685 - "pjsip show version" CLI command (Reported by
Joshua Colp)
* ASTERISK-24632 - install_prereq script installs pjproject
without IPv6 support (Reported by Rusty Newton)
* ASTERISK-24085 - Documentation - We should remove or further
document the 'contact' section in pjsip.conf (Reported by Rusty
Newton)
* ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported by
JoshE)
* ASTERISK-24700 - CRASH: NULL channel is being passed to
ast_bridge_transfer_attended() (Reported by Zane Conkle)
* ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
(Reported by Corey Farrell)
* ASTERISK-24799 - [patch] make fails with undefined reference to
SSLv3_client_method (Reported by Alexander Traud)
* ASTERISK-22670 - Asterisk crashes when processing ISDN AoC
Events (Reported by klaus3000)
* ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdn
call (Reported by Marcel Manz)
* ASTERISK-24740 - [patch]Segmentation fault on aoc-e event
(Reported by Panos Gkikakis)
* ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
for playing back messages stored in IMAP - play_message: No
origtime (Reported by Graham Barnett)
* ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
OSX with 64 bit integers (Reported by Corey Farrell)
* ASTERISK-24796 - Codecs and bucket schema's prevent module
unload (Reported by Corey Farrell)
* ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
(Reported by Ashley Sanders)
* ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
is invalid (Reported by Rusty Newton)
* ASTERISK-24785 - 'Expires' header missing from 200 OK on
REGISTER (Reported by Ross Beer)
* ASTERISK-24677 - ARI GET variable on channel provides unhelpful
response on non-existent variable (Reported by Joshua Colp)
* ASTERISK-24797 - bridge_softmix: G.729 codec license held
(Reported by Kevin Harwell)
* ASTERISK-24812 - ARI: Creating channels through /channels
resource always uses SLIN, which results in unneeded transcoding
(Reported by Matt Jordan)
* ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
thread ID being passed to pthread_kill (Reported by JoshE)
* ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
fail (Reported by Terry Wilson)
* ASTERISK-23214 - chan_sip WARNING message 'We are requesting
SRTP for audio, but they responded without it' is ambiguous and
wrong in some cases (Reported by Rusty Newton)
* ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
error response and BYE are sent to the caller (Reported by
Makoto Dei)
* ASTERISK-18105 - most of asterisk modules are unbuildable in
cygwin environment (Reported by feyfre)
* ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
* ASTERISK-24751 - Integer values in json payload to ARI cause
asterisk to crash (Reported by jeffrey putnam)
* ASTERISK-24838 - chan_sip: Locking inversion occurs when
building a peer causes a peer poke during request handling
(Reported by Richard Mudgett)
* ASTERISK-24825 - Caller ID not recognized using
Centrex/Distinctive dialing (Reported by Richard Mudgett)
* ASTERISK-24830 - res_rtp_asterisk.c checks USE_PJPROJECT not
HAVE_PJPROJECT (Reported by Stefan Engström)
* ASTERISK-24840 - res_pjsip: conflicting endpoint identifiers
(Reported by Kevin Harwell)
* ASTERISK-24755 - Asterisk sends unexpected early BYE to
transferrer during attended transfer when using a Stasis bridge
(Reported by John Bigelow)
* ASTERISK-24739 - [patch] - Out of files -- call fails --
numerous files with inodes from under /usr/share/zoneinfo,
mostly posixrules (Reported by Ed Hynan)
* ASTERISK-23390 - NewExten Event with application AGI shows up
before and after AGI runs (Reported by Benjamin Keith Ford)
* ASTERISK-24786 - [patch] - Asterisk terminates when playing a
voicemail stored in LDAP (Reported by Graham Barnett)
* ASTERISK-24808 - res_config_odbc: Improper escaping of
backslashes occurs with MySQL (Reported by Javier Acosta)
* ASTERISK-24807 - Missing mandatory field Max-Forwards (Reported
by Anatoli)
* ASTERISK-20850 - [patch]Nested functions aren't portable.
Adapting RAII_VAR to use clang/llvm blocks to get the
same/similar functionality. (Reported by Diederik de Groot)
* ASTERISK-24872 - [patch] AMI PJSIPShowEndpoint closes AMI
connection on error (Reported by Dmitriy Serov)
* ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
by Frank DiGennaro)
* ASTERISK-21038 - Bad command completion of "core set debug
channel" (Reported by Richard Kenner)
* ASTERISK-18708 - func_curl hangs channel under load (Reported by
Dave Cabot)
* ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
Atis Lezdins)
* ASTERISK-24876 - Investigate reference leaks from
tests/channels/local/local_optimize_away (Reported by Corey
Farrell)
* ASTERISK-24882 - chan_sip: Improve usage of REF_DEBUG (Reported
by Corey Farrell)
* ASTERISK-24817 - init_logger_chain: unreachable code block
(Reported by Corey Farrell)
* ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported by
snuffy)
* ASTERISK-24879 - [patch]Compilation fails due to 64bit time
under OpenBSD (Reported by snuffy)
Improvements made in this release:
-----------------------------------
* ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes
(Reported by Ben Merrills)
* ASTERISK-24811 - asterisk-publication sorcery object does not
use realtime (Reported by Matt Hoskins)
* ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
Couldn't find mailbox %s in context (Reported by Graham Barnett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0