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
Foto di gruppo a fine “Corso Asterisk 13 Avanzato”
Ringraziamo tutti i partecipanti al corso.
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.
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.