ASTERWEB Blog

31Ago/15Off

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

4Ago/15Off

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

4Ago/15Off

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

15Giu/15Off

Digium abbandona il progetto Asterisk-GUI

asterisk-gui-20-incoming-rules-e1434365309420-718x210

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.

6Giu/15Off

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

4Giu/15Off

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

6Mag/15Off

Rilasciato CLASS – Modulo beta – Monitoraggio Code in Realtime

Logo Asterweb

Logo Asterweb

Abbiamo rilasciato il modulo Code Monitoraggio in Realtime" in versione beta. Questo uno screenshot:
Gestione Code Realtime

Il rilascio stabile è previsto per il giorno 20 maggio con anche le statistiche sulle code.

Lo Staff Asterweb

9Apr/15Off

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

7Apr/15Off

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

2Apr/15Off

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

2Apr/15Off

Rilasciato Asterisk 11.17.0

Il giorno 01 aprile 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 11.17.0.

Dal post originale:
The release of Asterisk 11.17.0 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-17899 - Handle crypto lifetime in SDES-SRTP negotiation
(Reported by Dwayne Hubbard)

Bugs fixed in this release:
-----------------------------------
* ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
res_odbc (Reported by ibercom)
* ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
with replaces (Reported by Eelco Brolman)
* 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-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-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-24797 - bridge_softmix: G.729 codec license held
(Reported by Kevin Harwell)
* 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-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-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-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-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-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-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-11.17.0

8Feb/15Off

Rilasciato Asterisk 13.2.0

Il giorno 06 febbraio 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.2.0.

Dal post originale:
The release of Asterisk 13.2.0 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-24342 - PJSIP: Qualifying endpoints attempts to do them
all at the same time. (Reported by Richard Mudgett)
* ASTERISK-24514 - res_pjsip_outbound_registration: stack overflow
when using non-default sorcery wizard (Reported by Kevin
Harwell)
* ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
from JSSIP (Reported by Badalian Vyacheslav)
* ASTERISK-24607 - res_pjsip_session: re-INVITE with declined
media streams results in 488 (Reported by Matt Jordan)
* ASTERISK-24563 - Direct Media calls within private network
sometimes get one way audio (Reported by Kevin Harwell)
* ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to
race condition in accessing codec in stored ast_frame and codec
core (Reported by Matt Jordan)
* ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
enabled (Reported by Richard Mudgett)
* ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
enabled (Reported by Andreas Steinmetz)
* ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
casts char to unsigned int (Reported by Walter Doekes)
* ASTERISK-24536 - AMI redirect with PJSIP fails to move extra
channel (Reported by Niklas Larsson)
* ASTERISK-24459 - bridge_native_rtp: Native RTP bridging is
chosen for RTP compatible channels when the DTMF mode is not
compatible (Reported by Yaniv Simhi)
* ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
level - 'Remote address is null, most likely RTP has been
stopped' (Reported by Rusty Newton)
* ASTERISK-24513 - Local channel apparently leaked in off-nominal
DTMF attended transfer (Reported by Mark Michelson)
* ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
on startup (Reported by Richard Kenner)
* ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
destination when 'sendrpid=yes' (in proxy environment) (Reported
by Karsten Wemheuer)
* ASTERISK-23841 - DTMF atxfer doesn't set CallerID for the recall
calls to the transferrer. (Reported by Richard Mudgett)
* ASTERISK-24376 - res_pjsip_refer: REFER request for remote
session attempts to direct channel to external_replaces
extension instead of context, without providing for the
Referred-To SIP URI (Reported by Matt Jordan)
* ASTERISK-24591 - Stasis() side of an ARI originated channel
cannot be Redirected (Reported by Kinsey Moore)
* ASTERISK-24049 - Asterisk Manager Interface: A number of list
type responses aren't using astman_send_listack (Reported by
Jonathan Rose)
* ASTERISK-24637 - Channel re-enters Stasis() when it should not
(Reported by John Bigelow)
* ASTERISK-24474 - sip_to_pjsip.py lacks documentation and does
not function (Reported by John Kiniston)
* ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
(Reported by Kristian Høgh)
* ASTERISK-20744 - [patch] Security event logging does not work
over syslog (Reported by Michael Keuter)
* ASTERISK-24665 - Configure check required for
pjsip_get_dest_info() (Reported by Mark Michelson)
* ASTERISK-23850 - Park Application does not respect Return
Context Priority (Reported by Andrew Nagy)
* ASTERISK-23991 - [patch]asterisk.pc file contains a small error
in the CFlags returned (Reported by Diederik de Groot)
* ASTERISK-24655 - res_pjsip_outbound_publish: Hang on shutdown
while attempting to publish (Reported by Kevin Harwell)
* ASTERISK-24485 - res_pjsip cannot be unloaded or shutdown
(Reported by Corey Farrell)
* ASTERISK-24663 - [patch] Unnamed semaphore autoconf check fails
on cross compilation (Reported by abelbeck)
* ASTERISK-24624 - Transfer to invalid extension results in hung
channel. (Reported by Zane Conkle)
* ASTERISK-24615 - When Multiple Transports Exist in pjsip.conf,
Incorrect External Addresses is Used in SIP Packets When
Responding to INVITE (Reported by David Justl)
* ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
voicemail is not deleted after review, hangup (Reported by LEI
FU)
* ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
32-bit packages on 64-bit hosts (Reported by Ben Klang)
* ASTERISK-24600 - Stuck IAX channels, Asterisk stops responding
to most traffic, potential deadlock (Reported by Jeff Collell)
* ASTERISK-24560 - Creating a named ARI bridge twice causes a
crash (Reported by Kinsey Moore)
* ASTERISK-24682 - app_dial: Multiple DialEnd events emitted when
MACRO_RESULT or GOSUB_RESULT are an unexpected value (Reported
by Matt Jordan)
* ASTERISK-24640 - Registration pending stays forever after sip
reload (Reported by Max Man)
* ASTERISK-24673 - outgoing sip registers cannot be removed or
modified without doing restart (or doing module unload
chan_sip.so) (Reported by Stefan Engström)
* ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
m() option does not queue an MWI event (Reported by Gareth
Palmer)
* ASTERISK-24649 - Pushing of channel into bridge fails; Stasis
fails to get app name (Reported by John Bigelow)
* ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
column comparison for 'defaultuser' (Reported by
HZMI8gkCvPpom0tM)
* ASTERISK-24693 - Investigate and fix memory leaks in Asterisk
(Reported by Kevin Harwell)
* ASTERISK-24626 - Voicemail passwords not being stored in ARA
(Reported by Paddy Grice)
* ASTERISK-24539 - Compile fails on OSX because of sem_timedwait
in bridge_channel.c (Reported by George Joseph)
* ASTERISK-24544 - Compile fails on OSX Yosemite because of
incorrect detection of htonll and ntohll (Reported by George
Joseph)
* ASTERISK-24723 - confbridge: CLI command 'confbridge list XXXX'
no longer displays user menus (Reported by Matt Jordan)
* ASTERISK-24721 - manager: ModuleLoad action incorrectly reports
'module not found' during a Reload operation (Reported by Matt
Jordan)
* ASTERISK-24719 - ConfBridge recording channels get stuck when
recording started/stopped more than once (Reported by Richard
Mudgett)
* ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
by Kevin Harwell)
* ASTERISK-24728 - tcptls: Bad file descriptor error when
reloading chan_sip (Reported by Kevin Harwell)
* ASTERISK-24729 - Outbound registration not occuring on new
registrations after reload. (Reported by Richard Mudgett)
* ASTERISK-24676 - Security Vulnerability: URL request injection
in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
* ASTERISK-24666 - Security Vulnerability: RTP not closed after
sip call using unsupported codec (Reported by Y Ateya)
* ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
versions (Reported by Jared Biel)
* ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
Stephan Eisvogel)
* ASTERISK-24736 - Memory Leak Fixes (Reported by Mark Michelson)
* ASTERISK-24635 - PJSIP outbound PUBLISH crashes when no response
is ever received (Reported by Marco Paland)
* ASTERISK-24737 - When agent not logged in, agent status shows
unavailable, queue status shows agent invalid (Reported by
Richard Mudgett)

Improvements made in this release:
-----------------------------------
* ASTERISK-24552 - ARI: Allow associating a channel as an
initiator of an Origination for record keeping purposes
(Reported by Matt Jordan)
* ASTERISK-24553 - ARI/AMI: Include language in standard channel
snapshot output (Reported by Matt Jordan)
* ASTERISK-24643 - res_pjsip: Add user=phone option (Reported by
Matt Jordan)
* ASTERISK-24644 - res_pjsip_keepalive: Add keepalive module for
connection-oriented transports. (Reported by Matt Jordan)
* ASTERISK-24412 - [patch]Incomplete channel originate/continue
handling with ARI (Reported by Nir Simionovich (GreenfieldTech -
Israel))
* ASTERISK-24678 - [PATCH] Added atxfer* settings to
features.conf.sample (Reported by Niklas Larsson)
* ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported
by cloos)
* ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by
Dan Jenkins)
* ASTERISK-24316 - For httpd server, need option to define server
name for security purposes (Reported by Andrew Nagy)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.0

8Feb/15Off

Rilasciato Asterisk 11.16.0

Il giorno 06 febbraio 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 11.16.0.

Dal post originale:
The release of Asterisk 11.16.0 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-24472 - Asterisk Crash in OpenSSL when calling over WSS
from JSSIP (Reported by Badalian Vyacheslav)
* ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
enabled (Reported by Richard Mudgett)
* ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
enabled (Reported by Andreas Steinmetz)
* ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
casts char to unsigned int (Reported by Walter Doekes)
* ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
level - 'Remote address is null, most likely RTP has been
stopped' (Reported by Rusty Newton)
* ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
on startup (Reported by Richard Kenner)
* ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
destination when 'sendrpid=yes' (in proxy environment) (Reported
by Karsten Wemheuer)
* ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
(Reported by Kristian Høgh)
* ASTERISK-20744 - [patch] Security event logging does not work
over syslog (Reported by Michael Keuter)
* ASTERISK-23850 - Park Application does not respect Return
Context Priority (Reported by Andrew Nagy)
* ASTERISK-23991 - [patch]asterisk.pc file contains a small error
in the CFlags returned (Reported by Diederik de Groot)
* ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
voicemail is not deleted after review, hangup (Reported by LEI
FU)
* ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
32-bit packages on 64-bit hosts (Reported by Ben Klang)
* ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
m() option does not queue an MWI event (Reported by Gareth
Palmer)
* ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
column comparison for 'defaultuser' (Reported by
HZMI8gkCvPpom0tM)
* ASTERISK-24719 - ConfBridge recording channels get stuck when
recording started/stopped more than once (Reported by Richard
Mudgett)
* ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
by Kevin Harwell)
* ASTERISK-24728 - tcptls: Bad file descriptor error when
reloading chan_sip (Reported by Kevin Harwell)
* ASTERISK-24676 - Security Vulnerability: URL request injection
in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
* ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
versions (Reported by Jared Biel)
* ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
Stephan Eisvogel)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0

6Gen/15Off

Sangoma compra Schmooze e quindi FreePBX

Il giorno 1 gennaio 2015 Sangoma ha annunciato l'acquisto di Schmooze Com Inc. che è il principale sviluppatore e manager/sponsor del progetto Open Source FreePBX.

Per l'acquisizione di Schmooze Com Inc. Sangoma ha pagato cash 4 milioni di dollari.

Tony Lewis cofondatore e CEO di Schmooze ha dichiarato:

The FreePBX community should benefit from the project being backed by a larger, mature public company with much broader resources and over 30 years of experience in telecom and a long pedigree in open source. We are excited about the stability and credibility this adds to the project and I expect that FreePBX users will really appreciate it.

Speriamo in bene!

18Dic/14Off

Rilasciato Asterisk 11.15.0

Il giorno 15 dicembre 2014, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 12.8.0.

Dal post originale:
The Asterisk Development Team has announced the release of Asterisk 11.15.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.15.0 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-20127 - [Regression] Config.c config_text_file_load()
unescapes semicolons ("\;" -> ";") turning them into comments
(corruption) on rewrite of a config file (Reported by George
Joseph)
* ASTERISK-24307 - Unintentional memory retention in stringfields
(Reported by Etienne Lessard)
* ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
extra calls to ast_module_unref (Reported by Corey Farrell)
* ASTERISK-24504 - chan_console: Fix reference leaks to pvt
(Reported by Corey Farrell)
* ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
length exceeds 50 (roughly) national symbols (Reported by
Dmitriy Bubnov)
* ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
revision r227276 (Reported by Xavier Hienne)
* ASTERISK-20402 - Unable to cancel (features.conf) attended
transfer (Reported by Matt Riddell)
* ASTERISK-24505 - manager: http connections leak references
(Reported by Corey Farrell)
* ASTERISK-24502 - Build fails when dev-mode, dont optimize and
coverage are enabled (Reported by Corey Farrell)
* ASTERISK-24444 - PBX: Crash when generating extension for
pattern matching hint (Reported by Leandro Dardini)
* ASTERISK-24522 - ConfBridge: delay occurs between kicking all
endmarked users when last marked user leaves (Reported by Matt
Jordan)
* ASTERISK-15242 - transmit_refer leaks sip_refer structures
(Reported by David Woolley)
* ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
* ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
allow blocked addresses through (Reported by Matt Jordan)
* ASTERISK-24516 - [patch]Asterisk segfaults when playing back
voicemail under high concurrency with an IMAP backend (Reported
by David Duncan Ross Palmer)
* ASTERISK-24572 - [patch]App_meetme is loaded without its
defaults when the configuration file is missing (Reported by
Nuno Borges)
* ASTERISK-24573 - [patch]Out of sync conversation recording when
divided in multiple recordings (Reported by Nuno Borges)

Improvements made in this release:
-----------------------------------
* ASTERISK-24283 - [patch]Microseconds precision in the eventtime
column in the cel_odbc module (Reported by Etienne Lessard)
* ASTERISK-24530 - [patch] app_record stripping 1/4 second from
recordings (Reported by Ben Smithurst)
* ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
lookups (Reported by Birger "WIMPy" Harzenetter)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0