ASTERWEB Blog

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

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: