Rilasciato Asterisk 14.2.0
Il giorno 23 novembre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 14.2.0.
Dal post originale:
The release of Asterisk 14.2.0 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:
Improvements made in this release:
-----------------------------------
* ASTERISK-26558 - app_queue: add variable to know if the call is not answered after a queue (Reported by scgm11)
* ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry (Reported by scgm11)
* ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample (Reported by Kevin Harwell)
* ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands (Reported by Matt Jordan)
* ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP (Reported by Michael Walton)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy)
* ASTERISK-26520 - codec_opus: Generated fmtp line has no content (Reported by scgm11)
* ASTERISK-26605 - codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded. (Reported by Richard Mudgett)
* ASTERISK-26516 - pjsip: Memory corruption with possible memory leak. (Reported by Richard Mudgett)
* ASTERISK-26556 - manager: AMI version report same in Ast 13 & 14, despite Ast 14 syntax changes (Reported by Michelle Dupuis)
* ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss)
* ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage (Reported by George Joseph)
* ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set (Reported by Jason)
* ASTERISK-26575 - testsuite: Need to check PJSIP functionality when res_srtp is not loaded. (Reported by Joshua Colp)
* ASTERISK-26571 - res_pjsip: Resolution incorrect when explicit IPv6 transport configured (Reported by Joshua Colp)
* ASTERISK-26468 - ari: Bridge events stop working after this sequence of ARI calls (Reported by Daniele Pallastrelli)
* ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by Dmitry Melekhov)
* ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11 regressions (Reported by Matt Jordan)
* ASTERISK-26412 - build: Prepare for gcc 6.2 (Reported by George Joseph)
* ASTERISK-26509 - A few non-critical deprecation warnings when building on Ubuntu 16.10 (Reported by Jonathan Harris)
* ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter)
* ASTERISK-26549 - app_dial: When PickupChan() is used some channels may have incorrect device state (Reported by Joshua Colp)
* ASTERISK-24274 - [patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used (Reported by Frankie Chin)
* ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic payload types. (Reported by Alexander Traud)
* ASTERISK-26506 - [patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c (Reported by Matt Krokosz)
* ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele Giacone)
* ASTERISK-26476 - chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings" (Reported by Sergey Grachev)
* ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats to maximum (Reported by Joshua Colp)
* ASTERISK-26537 - AMI: NewConnectedLine event is not documented (Reported by Etienne Lessard)
* ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy (Reported by Badalian Vyacheslav)
* ASTERISK-26524 - astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled. (Reported by Corey Farrell)
* ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash (Reported by Ian Gilmour)
* ASTERISK-26387 - Asterisk segfaults shortly after starting even with no active calls. (Reported by Harley Peters)
* ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance (Reported by Joshua Colp)
* ASTERISK-26514 - Super Awesome Company: Don't specify transport in pjsip.conf (Reported by Rusty Newton)
* ASTERISK-26510 - pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions (Reported by George Joseph)
* ASTERISK-22480 - Embedded pjproject: build.mak contains hardcoded full path to version.mak (Reported by Matt Jordan)
* ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change (Reported by Bill Brigden)
* ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle)
* ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness (Reported by Andreas Wetzel)
* ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations. (Reported by Alexander Traud)
* ASTERISK-26482 - [patch] chan_pjsip: segfault on already disconnected session (Reported by Alexei Gradinari)
* ASTERISK-26421 - Segmentation Fault with ARI originate into mixing bridge with 43 clients (Reported by Andrew Nagy)
* ASTERISK-26444 - 'features show' command in CLI does not return prompt. (Reported by John Kiniston)
* ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module (Reported by Alexander Traud)
* ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir Cohen)
* ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage (Reported by Leandro Dardini)
* ASTERISK-26439 - chan_rtp: Crash when originating (Reported by Kayode)
* ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered. (Reported by Alexander Traud)
* ASTERISK-26618 - build: Backport addition of librt check to configure.ac (Reported by Kevin Harwell)
New Features made in this release:
-----------------------------------
* ASTERISK-26595 - ARI: Add the ability to control the source of video in a multi-party mixing bridge (Reported by Matt Jordan)
* ASTERISK-26492 - ARI: Add ability to specify channel variables on websocket events (Reported by Mark Michelson)
* ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing events (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-14.2.0
Rilasciato Asterisk 13.13.0
Il giorno 23 novembre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.13.0.
Dal post originale:
The release of Asterisk 13.13.0 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-26595 - ARI: Add the ability to control the source of video in a multi-party mixing bridge (Reported by Matt Jordan)
* ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing events (Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy)
* ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss)
* ASTERISK-26520 - codec_opus: Generated fmtp line has no content (Reported by scgm11)
* ASTERISK-26605 - codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded. (Reported by Richard Mudgett)
* ASTERISK-26516 - pjsip: Memory corruption with possible memory leak. (Reported by Richard Mudgett)
* ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage (Reported by George Joseph)
* ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set (Reported by Jason)
* ASTERISK-26575 - testsuite: Need to check PJSIP functionality when res_srtp is not loaded. (Reported by Joshua Colp)
* ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by Dmitry Melekhov)
* ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11 regressions (Reported by Matt Jordan)
* ASTERISK-26412 - build: Prepare for gcc 6.2 (Reported by George Joseph)
* ASTERISK-26509 - A few non-critical deprecation warnings when building on Ubuntu 16.10 (Reported by Jonathan Harris)
* ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter)
* ASTERISK-26468 - ari: Bridge events stop working after this sequence of ARI calls (Reported by Daniele Pallastrelli)
* ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic payload types. (Reported by Alexander Traud)
* ASTERISK-26549 - app_dial: When PickupChan() is used some channels may have incorrect device state (Reported by Joshua Colp)
* ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats to maximum (Reported by Joshua Colp)
* ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele Giacone)
* ASTERISK-26476 - chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings" (Reported by Sergey Grachev)
* ASTERISK-26537 - AMI: NewConnectedLine event is not documented (Reported by Etienne Lessard)
* ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy (Reported by Badalian Vyacheslav)
* ASTERISK-26524 - astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled. (Reported by Corey Farrell)
* ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash (Reported by Ian Gilmour)
* ASTERISK-26387 - Asterisk segfaults shortly after starting even with no active calls. (Reported by Harley Peters)
* ASTERISK-26514 - Super Awesome Company: Don't specify transport in pjsip.conf (Reported by Rusty Newton)
* ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance (Reported by Joshua Colp)
* ASTERISK-26510 - pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions (Reported by George Joseph)
* ASTERISK-22480 - Embedded pjproject: build.mak contains hardcoded full path to version.mak (Reported by Matt Jordan)
* ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change (Reported by Bill Brigden)
* ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle)
* ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness (Reported by Andreas Wetzel)
* ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations. (Reported by Alexander Traud)
* ASTERISK-26421 - Segmentation Fault with ARI originate into mixing bridge with 43 clients (Reported by Andrew Nagy)
* ASTERISK-26444 - 'features show' command in CLI does not return prompt. (Reported by John Kiniston)
* ASTERISK-26482 - [patch] chan_pjsip: segfault on already disconnected session (Reported by Alexei Gradinari)
* ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module (Reported by Alexander Traud)
* ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir Cohen)
* ASTERISK-26439 - chan_rtp: Crash when originating (Reported by Kayode)
* ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage (Reported by Leandro Dardini)
* ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered. (Reported by Alexander Traud)
* ASTERISK-26618 - build: Backport addition of librt check to configure.ac (Reported by Kevin Harwell)
Improvements made in this release:
-----------------------------------
* ASTERISK-25063 - [patch]add X.509 subject alternative name support to Asterisk TLS support (Reported by Maciej Szmigiero)
* ASTERISK-26558 - app_queue: add variable to know if the call is not answered after a queue (Reported by scgm11)
* ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry (Reported by scgm11)
* ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample (Reported by Kevin Harwell)
* ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands (Reported by Matt Jordan)
* ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP (Reported by Michael Walton)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.13.0
Rilasciato Asterisk 11.25.0
Il giorno 23 novembre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 11.25.0.
Dal post originale:
The release of Asterisk 11.25.0 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-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle)
* ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module (Reported by Alexander Traud)
* ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir Cohen)
* ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage (Reported by Leandro Dardini)
* ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered. (Reported by Alexander Traud)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
Rilasciato Asterisk 14.2.0-rc2
Il giorno 22 novembre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 14.2.0-rc2.
Dal post originale:
The release of Asterisk 14.2.0-rc2 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-26618 - build: Backport addition of librt check to
configure.ac (Reported by Kevin Harwell)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.2.0-rc2
Rilasciato Asterisk 13.13.0-rc2
Il giorno 22 novembre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.13.0-rc2.
Dal post originale:
The release of Asterisk 13.13.0-rc2 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-26618 - build: Backport addition of librt check to
configure.ac (Reported by Kevin Harwell)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.13.0-rc2
Rilasciato Asterisk 13.12.2
Il giorno 10 novembre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.12.2.
Dal post originale:
The release of Asterisk 13.12.2 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-13.12.2
Nuovo “Corso Asterisk 13 Avanzato” a Milano nei giorni 24-25-26 gennaio 2017
Nuovo "Corso Asterisk 13 Avanzato" a Milano nei giorni 24-25-26 gennaio 2017.
Sono aperte le iscrizioni al costo promozionale di € 390,00 più iva fino al 30/11/2016.
Vi aspettiamo.
Saluti, lo Staff
Foto di gruppo a fine “Corso Asterisk 13 Avanzato”
Un ringraziamento ed un "in bocca al lupo" ai partecipanti al corso.
Rilasciato Asterisk 14.1.1
Il giorno 27 ottobre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 14.1.1.
Dal post originale:
The release of Asterisk 14.1.1 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.1
Rilasciato Asterisk 13.12.1
Il giorno 27 ottobre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.12.1.
Dal post originale:
The release of Asterisk 13.12.1 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.1
Rilasciato Asterisk 11.24.1
Il giorno 27 ottobre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 11.24.1.
Dal post originale:
The release of Asterisk 11.24.1 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
Rilasciato Asterisk 14.1.0
Il giorno 25 ottobre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 14.1.0.
Dal post originale:
The release of Asterisk 14.1.0 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-26277 - Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment (Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26477 - pjproject: SEGV during SSL operations (Reported by George Joseph)
* ASTERISK-17470 - [patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio. (Reported by effie mouzeli)
* ASTERISK-26416 - pjproject-bundled: configure fails to check for all required utilities (Reported by Corey Farrell)
* ASTERISK-26466 - core: Be forgiving on external callerid that may be flawed so we don't drop events (Reported by Richard Mudgett)
* ASTERISK-26362 - res_config_mysql: Broken after 13.10 (Reported by Carlos Chavez)
* ASTERISK-26446 - app_dial: There's no way to override the hangupcause on unanswered channels (Reported by George Joseph)
* ASTERISK-26410 - core: Asterisk 14 doesn't show the header in the console or verbose when starting (Reported by Dan Jenkins)
* ASTERISK-24311 - Populating database via Alembic fails when using same database for multiple schema sets (Reported by Dafi Ni)
* ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response. (Reported by Alexander Traud)
* ASTERISK-26426 - format_ogg_opus: remove from source (Reported by Kevin Harwell)
* ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::] (Reported by Jacek)
* ASTERISK-25468 - Deadlock in chan_sip - core show locks shows do_monitor lock (Reported by Barry Flanagan)
* ASTERISK-26397 - manager: PresenceState action crashes Asterisk 14 (Reported by Andrew Nagy)
* ASTERISK-26389 - res_odbc: Clean up pooling options (Reported by Joshua Colp)
* ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed (Reported by Tzafrir Cohen)
* ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled (Reported by Anthony Messina)
* ASTERISK-26391 - Consoles do not display verbose logger messages even when requested. (Reported by Marcelo Terres)
* ASTERISK-26263 - SQL error when using realtime and registering extension / inserting into ps_contacts (Reported by Jeppe Ryskov Larsen)
* ASTERISK-26365 - rtp: Offer with multiple payloads for same codec is incorrectly handled (Reported by Joshua Colp)
* ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron Hamstra)
* ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten for connectionful protocols (Reported by Joshua Colp)
* ASTERISK-26367 - rtp: Timestamps broken when video frame is across multiple RTP packets (Reported by Joshua Colp)
* ASTERISK-26375 - res_pjsip_transport_management: Log message states seconds, but time value is milliseconds (Reported by Joshua Colp)
* ASTERISK-26364 - res_pjsip: Don't assume a request will have target addresses (Reported by Joshua Colp)
* ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs. (Reported by Richard Mudgett)
* ASTERISK-26316 - res_pjsip_callerid: Irregular URI causes unexpected callerid (Reported by Kevin Harwell)
* ASTERISK-26349 - 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed (Reported by Dmitry Melekhov)
* ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by Etienne Lessard)
* ASTERISK-26264 - res_pjsip: Crash when applying ACL from non-existent endpoint (Reported by nappsoft)
* ASTERISK-26341 - ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list (Reported by Matt Jordan)
* ASTERISK-25691 - Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up. (Reported by Etienne Lessard)
* ASTERISK-26331 - Crash on “core show channeltype Surrogate†in ast_format_cap_get_names (Reported by CGI.NET)
* ASTERISK-26269 - res_pjsip: Wrong state for aors without registered contacts after startup (Reported by nappsoft)
* ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's' extension" (Reported by chris de rock)
* ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts (Reported by Etienne Lessard)
* ASTERISK-26299 - app_queue: Queue application sometimes stops calling members with Local interface (Reported by Etienne Lessard)
* ASTERISK-26279 - pjproject-bundled: Fails to compile on Debian 6 (Reported by George Joseph)
* ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up properly (Reported by Alexander Traud)
* ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels (Reported by Etienne Lessard)
* ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code (Reported by David Brillert)
* ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels. (Reported by Richard Mudgett)
* ASTERISK-26288 - followme: fails to reset config items to default values on reload (Reported by Tzafrir Cohen)
* ASTERISK-22374 - Finish mapping the sip.conf parameters to res_sip.conf parameters (Reported by Matt Jordan)
* ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) (Reported by abelbeck)
* ASTERISK-26228 - res_pjsip_sdp_rtp: G729A does not include annexb=no attribute. (Reported by Ali Ghavidel)
* ASTERISK-25472 - Swagger scripts are not replacing format variable in file brief (Reported by Corey Farrell)
* ASTERISK-25984 - res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it (Reported by József Dudás)
* ASTERISK-26305 - Asterisk 14: Two resolver unbound testsuite tests fail (Reported by Richard Mudgett)
* ASTERISK-26303 - [patch] BuildSystem: ca_list_path capabilities not detected in PJProject. (Reported by Alexander Traud)
* ASTERISK-25492 - ARI: Path parameters are case sensitive (Reported by Joshua Colp)
* ASTERISK-26164 - XMPP no longer triggers NOTIFY to device via chan_pjsip (Reported by Ross Beer)
* ASTERISK-26233 - pbx: Failure to remove inconsistent extension names (Reported by Corey Farrell)
* ASTERISK-26246 - Security: Privilege escalation by AMI adding dialplan extensions. (Reported by Richard Mudgett)
* ASTERISK-26267 - ast_register_atexit callbacks should be run on failed startup. (Reported by Corey Farrell)
* ASTERISK-26241 - res_pjsip: When using compact headers, rpid and pai are incorrectly generated (Reported by George Joseph)
* ASTERISK-25797 - app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension (Reported by Etienne Lessard)
* ASTERISK-26239 - res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname (Reported by George Joseph)
* ASTERISK-26238 - res_pjsip: Empty global default_from_user causes crash (Reported by Joshua Colp)
* ASTERISK-26268 - alembic: 'auth_username' not in PJSIP 'identify_by' enum (Reported by Joshua Colp)
* ASTERISK-26253 - sdp_srtp: libsrtp now a required dependency, shouldn't be (Reported by Ben Merrills)
* ASTERISK-26145 - pjsip: Deadlock with suspend + masquerade + indicate (Reported by Ross Beer)
* ASTERISK-26183 - alembic: error when using sqlalchemy version 1.1.0b2 (Reported by Kevin Harwell)
* ASTERISK-26283 - res_resolver_unbound: fails configure on older Ubuntu and CentOS (Reported by George Joseph)
* ASTERISK-26280 - DNS lookups can block channel media paths (Reported by Mark Michelson)
* ASTERISK-26278 - asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM. (Reported by Corey Farrell)
* ASTERISK-25217 - [patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c (Reported by Richard Mudgett)
* ASTERISK-26265 - Errors ignored from some parts of system initialization. (Reported by Corey Farrell)
* ASTERISK-26206 - [patch] res_pjsip: Use more compatible regex for get all (Reported by Dmitry)
* ASTERISK-26256 - [patch] SIP/SDP origin (o=) contains brackets with IP6 (Reported by Alexander Traud)
* ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported by Andrew Nagy)
* ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..." (Reported by Hans van Eijsden)
* ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard Mudgett)
Improvements made in this release:
-----------------------------------
* ASTERISK-26409 - codec_opus: Update Asterisk to support the translation codec. (Reported by Kevin Harwell)
* ASTERISK-26289 - Announcer channels in ConfBridges cause inefficiencies (Reported by Mark Michelson)
* ASTERISK-25980 - [patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used (Reported by Alexei Gradinari)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.0
Rilasciato Asterisk 13.12.0
Il giorno 25 ottobre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.12.0.
Dal post originale:
The release of Asterisk 13.12.0 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-26277 - Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment (Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26477 - pjproject: SEGV during SSL operations (Reported by George Joseph)
* ASTERISK-17470 - [patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio. (Reported by effie mouzeli)
* ASTERISK-26416 - pjproject-bundled: configure fails to check for all required utilities (Reported by Corey Farrell)
* ASTERISK-26466 - core: Be forgiving on external callerid that may be flawed so we don't drop events (Reported by Richard Mudgett)
* ASTERISK-26362 - res_config_mysql: Broken after 13.10 (Reported by Carlos Chavez)
* ASTERISK-26446 - app_dial: There's no way to override the hangupcause on unanswered channels (Reported by George Joseph)
* ASTERISK-24311 - Populating database via Alembic fails when using same database for multiple schema sets (Reported by Dafi Ni)
* ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response. (Reported by Alexander Traud)
* ASTERISK-26426 - format_ogg_opus: remove from source (Reported by Kevin Harwell)
* ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::] (Reported by Jacek)
* ASTERISK-25468 - Deadlock in chan_sip - core show locks shows do_monitor lock (Reported by Barry Flanagan)
* ASTERISK-26397 - manager: PresenceState action crashes Asterisk 14 (Reported by Andrew Nagy)
* ASTERISK-26389 - res_odbc: Clean up pooling options (Reported by Joshua Colp)
* ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed (Reported by Tzafrir Cohen)
* ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled (Reported by Anthony Messina)
* ASTERISK-26263 - SQL error when using realtime and registering extension / inserting into ps_contacts (Reported by Jeppe Ryskov Larsen)
* ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten for connectionful protocols (Reported by Joshua Colp)
* ASTERISK-26367 - rtp: Timestamps broken when video frame is across multiple RTP packets (Reported by Joshua Colp)
* ASTERISK-26375 - res_pjsip_transport_management: Log message states seconds, but time value is milliseconds (Reported by Joshua Colp)
* ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron Hamstra)
* ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs. (Reported by Richard Mudgett)
* ASTERISK-26316 - res_pjsip_callerid: Irregular URI causes unexpected callerid (Reported by Kevin Harwell)
* ASTERISK-26349 - 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed (Reported by Dmitry Melekhov)
* ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by Etienne Lessard)
* ASTERISK-26264 - res_pjsip: Crash when applying ACL from non-existent endpoint (Reported by nappsoft)
* ASTERISK-26288 - followme: fails to reset config items to default values on reload (Reported by Tzafrir Cohen)
* ASTERISK-25691 - Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up. (Reported by Etienne Lessard)
* ASTERISK-26331 - Crash on “core show channeltype Surrogate†in ast_format_cap_get_names (Reported by CGI.NET)
* ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's' extension" (Reported by chris de rock)
* ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts (Reported by Etienne Lessard)
* ASTERISK-26279 - pjproject-bundled: Fails to compile on Debian 6 (Reported by George Joseph)
* ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up properly (Reported by Alexander Traud)
* ASTERISK-26299 - app_queue: Queue application sometimes stops calling members with Local interface (Reported by Etienne Lessard)
* ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels (Reported by Etienne Lessard)
* ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code (Reported by David Brillert)
* ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels. (Reported by Richard Mudgett)
* ASTERISK-26269 - res_pjsip: Wrong state for aors without registered contacts after startup (Reported by nappsoft)
* ASTERISK-22374 - Finish mapping the sip.conf parameters to res_sip.conf parameters (Reported by Matt Jordan)
* ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) (Reported by abelbeck)
* ASTERISK-25472 - Swagger scripts are not replacing format variable in file brief (Reported by Corey Farrell)
* ASTERISK-26228 - res_pjsip_sdp_rtp: G729A does not include annexb=no attribute. (Reported by Ali Ghavidel)
* ASTERISK-25984 - res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it (Reported by József Dudás)
* ASTERISK-26305 - Asterisk 14: Two resolver unbound testsuite tests fail (Reported by Richard Mudgett)
* ASTERISK-26303 - [patch] BuildSystem: ca_list_path capabilities not detected in PJProject. (Reported by Alexander Traud)
* ASTERISK-25492 - ARI: Path parameters are case sensitive (Reported by Joshua Colp)
* ASTERISK-26233 - pbx: Failure to remove inconsistent extension names (Reported by Corey Farrell)
* ASTERISK-26164 - XMPP no longer triggers NOTIFY to device via chan_pjsip (Reported by Ross Beer)
* ASTERISK-26246 - Security: Privilege escalation by AMI adding dialplan extensions. (Reported by Richard Mudgett)
* ASTERISK-26267 - ast_register_atexit callbacks should be run on failed startup. (Reported by Corey Farrell)
* ASTERISK-26241 - res_pjsip: When using compact headers, rpid and pai are incorrectly generated (Reported by George Joseph)
* ASTERISK-26239 - res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname (Reported by George Joseph)
* ASTERISK-26238 - res_pjsip: Empty global default_from_user causes crash (Reported by Joshua Colp)
* ASTERISK-25797 - app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension (Reported by Etienne Lessard)
* ASTERISK-26268 - alembic: 'auth_username' not in PJSIP 'identify_by' enum (Reported by Joshua Colp)
* ASTERISK-26145 - pjsip: Deadlock with suspend + masquerade + indicate (Reported by Ross Beer)
* ASTERISK-26183 - alembic: error when using sqlalchemy version 1.1.0b2 (Reported by Kevin Harwell)
* ASTERISK-26280 - DNS lookups can block channel media paths (Reported by Mark Michelson)
* ASTERISK-25217 - [patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as
res_pjsip_outbound_registration.c (Reported by Richard Mudgett)
* ASTERISK-26265 - Errors ignored from some parts of system initialization. (Reported by Corey Farrell)
* ASTERISK-26206 - [patch] res_pjsip: Use more compatible regex for get all (Reported by Dmitry)
* ASTERISK-26256 - [patch] SIP/SDP origin (o=) contains brackets with IP6 (Reported by Alexander Traud)
* ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported by Andrew Nagy)
* ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..." (Reported by Hans van Eijsden)
Improvements made in this release:
-----------------------------------
* ASTERISK-26409 - codec_opus: Update Asterisk to support the translation codec. (Reported by Kevin Harwell)
* ASTERISK-26289 - Announcer channels in ConfBridges cause inefficiencies (Reported by Mark Michelson)
* ASTERISK-25980 - [patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used (Reported by Alexei Gradinari)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.0
Rilasciato Asterisk 11.24.0
Il giorno 25 ottobre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 11.24.0.
Dal post originale:
The Asterisk Development Team has announced the release of Asterisk 11.24.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.24.0 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-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response. (Reported by Alexander Traud)
* ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::] (Reported by Jacek)
* ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed (Reported by Tzafrir Cohen)
* ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron Hamstra)
* ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs. (Reported by Richard Mudgett)
* ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by Etienne Lessard)
* ASTERISK-26288 - followme: fails to reset config items to default values on reload (Reported by Tzafrir Cohen)
* ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's' extension" (Reported by chris de rock)
* ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts (Reported by Etienne Lessard)
* ASTERISK-26299 - app_queue: Queue application sometimes stops calling members with Local interface (Reported by Etienne Lessard)
* ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up properly (Reported by Alexander Traud)
* ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels (Reported by Etienne Lessard)
* ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code (Reported by David Brillert)
* ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels. (Reported by Richard Mudgett)
* ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have multiple native formats (Reported by Matt Jordan)
* ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) (Reported by abelbeck)
* ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change) (Reported by Krzysztof Trempala)
* ASTERISK-26233 - pbx: Failure to remove inconsistent extension names (Reported by Corey Farrell)
* ASTERISK-26267 - ast_register_atexit callbacks should be run on failed startup. (Reported by Corey Farrell)
* ASTERISK-26265 - Errors ignored from some parts of system initialization. (Reported by Corey Farrell)
* ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported by Andrew Nagy)
* ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard Mudgett)
* ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and attended transfer handling (Reported by Ben Smithurst)
* ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional code. (Reported by Corey Farrell)
* ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even on lost packets. (Reported by Alexander Traud)
* ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files (Reported by Tzafrir Cohen)
* ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by Richard Miller)
* ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported by Corey Farrell)
* ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure occurred on RTP instance (Reported by Edwin Vandamme)
* ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by Alexander Traud)
* ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when building menuselect (Reported by Jeffrey Walton)
* ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and bugs (Reported by Alexei Gradinari)
* ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua Colp)
* ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George Joseph)
Improvements made in this release:
-----------------------------------
* ASTERISK-26220 - Add support for noreturn function attributes. (Reported by Corey Farrell)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
Asterweb e Snom Italia – FreeWebinar – SIP e NAT: problemi e soluzioni
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Asterweb, di concerto con Snom Italia, organizza questo Free-Webinar che riteniamo possa essere di interesse per molti operatori del settore.
Relatori:
- Gaspare Noto di Asterweb Srl
- Luca Livraga di Snom Italia
INFO:
Da martedì 4 ottobre 2016 inizieremo la pubblicazione di una serie di tutorials sui prodotti Snom che potrete trovare nella sezione "Tutorials" del sito asterweb.
DATA: 08 NOVEMBRE 2016
Inizio: ore 14:30
Durata: 30/45 minuti
Costo: Nessuno