ASTERWEB Blog

28Ott/16Off

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

28Ott/16Off

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

28Ott/16Off

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

27Ott/16Off

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

27Ott/16Off

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

27Ott/16Off

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

28Set/16Off

Asterweb e Snom Italia – FreeWebinar – SIP e NAT: problemi e soluzioni

logo-asterweb logo-snom_gray

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

10Set/16Off

Rilasciato Asterisk 13.11.2

Il giorno 09 settembre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk Asterisk 13.11.2.

Dal post originale:

The release of Asterisk 13.11.2 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-26349 - 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed (Reported by Dmitry Melekhov)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.11.2

9Set/16Off

AST-2016-007: RTP Resource Exhaustion

Il giorno 08 settembre 2016, l'Asterisk Security Team ha rilasciato il seguente post.

Dal post originale:

               Asterisk Project Security Advisory - AST-2016-007

Product Asterisk
Summary RTP Resource Exhaustion
Nature of Advisory Denial of Service
Susceptibility Remote Authenticated Sessions
Severity Moderate
Exploits Known No
Reported On August 5, 2016
Reported By Etienne Lessard
Posted On
Last Updated On September 8, 2016
Advisory Contact Joshua Colp <jcolp AT digium DOT com>
CVE Name

Description The overlap dialing feature in chan_sip allows chan_sip to
report to a device that the number that has been dialed is
incomplete and more digits are required. If this
functionality is used with a device that has performed
username/password authentication RTP resources are leaked.
This occurs because the code fails to release the old RTP
resources before allocating new ones in this scenario. If
all resources are used then RTP port exhaustion will occur
and no RTP sessions are able to be set up.

Resolution If overlap dialing support is not needed the “allowoverlapâ€
option can be set to no. This will stop any usage of the
scenario which causes the resource exhaustion.

If overlap dialing support is needed a change has been made
so that existing RTP resources are destroyed in this
scenario before allocating new resources.

Affected Versions
Product Release
Series
Asterisk Open Source 11.x All Versions
Asterisk Open Source 13.x All Versions
Certified Asterisk 11.6 All Versions
Certified Asterisk 13.8 All Versions

Corrected In
Product Release
Asterisk Open Source 11.23.1, 13.11.1
Certified Asterisk 11.6-cert15, 13.8-cert3

Patches
SVN URL Revision

Links https://issues.asterisk.org/jira/browse/ASTERISK-26272

Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security

This document may be superseded by later versions; if so, the latest
version will be posted at
http://downloads.digium.com/pub/security/AST-2016-007.pdf and
http://downloads.digium.com/pub/security/AST-2016-007.html

Revision History
Date Editor Revisions Made
August 23, 2016 Joshua Colp Initial creation

Asterisk Project Security Advisory - AST-2016-007
Copyright © 2016 Digium, Inc. All Rights Reserved.
Permission is hereby granted to distribute and publish this advisory in its
original, unaltered form.

9Set/16Off

AST-2016-006: Crash on ACK from unknown endpoint

Il giorno 08 settembre 2016, l'Asterisk Security Team ha rilasciato il seguente post.

Dal post originale:

               Asterisk Project Security Advisory - AST-2016-006

Product Asterisk
Summary Crash on ACK from unknown endpoint
Nature of Advisory Remote Crash
Susceptibility Remote unauthenticated sessions
Severity Critical
Exploits Known No
Reported On August 3, 2016
Reported By Nappsoft
Posted On
Last Updated On August 31, 2016
Advisory Contact mark DOT michelson AT digium DOT com
CVE Name

Description Asterisk can be crashed remotely by sending an ACK to it
from an endpoint username that Asterisk does not recognize.
Most SIP request types result in an "artificial" endpoint
being looked up, but ACKs bypass this lookup. The resulting
NULL pointer results in a crash when attempting to
determine if ACLs should be applied.

This issue was introduced in the Asterisk 13.10 release and
only affects that release.

This issue only affects users using the PJSIP stack with
Asterisk. Those users that use chan_sip are unaffected.

Resolution ACKs now result in an artificial endpoint being looked up
just like other SIP request types.

Affected Versions
Product Release
Series
Asterisk Open Source 11.x Unaffected
Asterisk Open Source 13.x 13.10.0
Certified Asterisk 11.6 Unaffected
Certified Asterisk 13.8 Unaffected

Corrected In
Product Release
Asterisk Open Source 13.11.1

Patches
SVN URL Revision

Links

Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security

This document may be superseded by later versions; if so, the latest
version will be posted at
http://downloads.digium.com/pub/security/AST-2016-006.pdf and
http://downloads.digium.com/pub/security/AST-2016-006.html

Revision History
Date Editor Revisions Made
August 16, 2016 Mark Michelson Initial draft of Advisory

Asterisk Project Security Advisory - AST-2016-006
Copyright (c) 2016 Digium, Inc. All Rights Reserved.
Permission is hereby granted to distribute and publish this advisory in its
original, unaltered form.

2Set/16Off

Rilasciato Asterisk 13.11.0

Il giorno 01 settembre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk Asterisk 13.11.0.

Dal post originale:

The release of Asterisk 13.11.0 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-25904 - PJSIP: add contact.updated event (Reported by Alexei Gradinari)

Bugs fixed in this release:
-----------------------------------
* ASTERISK-26269 - res_pjsip: Wrong state for aors without registered contacts after startup (Reported by nappsoft)
* ASTERISK-26299 - app_queue: Queue application sometimes stops calling members with Local interface (Reported by Etienne Lessard)
* 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)
* ASTERISK-26227 - sqlalchemy error due to long identifier name (Reported by Mark Michelson)
* ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron Hamstra)
* ASTERISK-26214 - Allow arbitrary time for fax detection to end on a channel (Reported by Richard Mudgett)
* ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and attended transfer handling (Reported by Ben Smithurst)
* ASTERISK-26216 - res_fax: Deadlock when detect fax while channel executing Playback (Reported by Richard Mudgett)
* ASTERISK-26212 - [patch] Makefile: Retain XML Declaration and DTD in docs. (Reported by Alexander Traud)
* 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-26200 - [patch] res_pjsip_mwi: improve realtime performance - remove unneeded check on endpoint's contacts. (Reported by Alexei Gradinari)
* 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-26193 - chan_sip: reference leak in mwi_event_cb (Reported by Corey Farrell)
* ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure occurred on RTP instance (Reported by Edwin Vandamme)
* ASTERISK-26191 - threadpool: Leak on duplicate taskprocessor for ast_threadpool_serializer_group (Reported by Corey Farrell)
* ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by Alexander Traud)
* ASTERISK-26160 - pjsip: Updated->Reachable during qualify (Reported by Matt Jordan)
* 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-26177 - func_odbc: Database handle is kept when it should be released (Reported by Leandro Dardini)
* ASTERISK-26184 - chan_sip: Reference leaks in error paths. (Reported by Corey Farrell)
* ASTERISK-26181 - REF_DEBUG: Node object incorrectly logged during duplicate replacement (Reported by Corey Farrell)
* ASTERISK-26180 - PJSIP: provide valid tcp nodelay option for reuse (Reported by Scott Griepentrog)
* ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua Colp)
* ASTERISK-26172 - res_sorcery_realtime: fix bug when successful sql UPDATE is treated as failed if there is no affected rows. (Reported by Alexei Gradinari)
* ASTERISK-25772 - res_pjsip: Unexpected two BYE when answered (Reported by Dmitriy Serov)
* ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request due to server timeout (Reported by Ross Beer)
* ASTERISK-26144 - Crash on loading codecs g729/g723 (Reported by Alexei Gradinari)
* ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George Joseph)
* ASTERISK-26021 - Build codecs siren7 and siren14 for Asterisk 13 (Reported by Daniel Denson)
* ASTERISK-26326 - Crash when dialing MulticastRTP channel (Reported by George Joseph)

Improvements made in this release:
-----------------------------------
* ASTERISK-26220 - Add support for noreturn function attributes. (Reported by Corey Farrell)
* ASTERISK-22131 - Update the make dependencies script to pull, build, and install the correct pjproject (Reported by Matt Jordan)
* ASTERISK-25471 - [patch]Add subscribe_context to res_pjsip (Reported by JoshE)
* ASTERISK-26159 - res_hep: enabled by default and information sent to default address (Reported by Ross Beer)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.11.0

30Ago/16Off

Rilasciato Asterisk 11.6-cert14

Il giorno 29 agosto 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk Asterisk 11.6-cert14.

Dal post originale:

The release of Certified Asterisk 11.6-cert14 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-23013 - [patch] Deadlock between 'sip show channels' command and attended transfer handling (Reported by Ben Smithurst)
* ASTERISK-25494 - build: GCC 5.1.x catches some new const, array bounds and missing paren issues (Reported by George Joseph)
* ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0 (Reported by Jeffrey C. Ollie)
* ASTERISK-26138 - chan_unistim: Under FreeBSD, chan_unistim generates a compile error (Reported by George Joseph)
* ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George Joseph)
* ASTERISK-23509 - [patch]SayNumber for Polish language tries to play empty files for numbers divisible by 100 (Reported by zvision)
* ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported by George Joseph)
* ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0 (Reported by Patrick Laimbock)
* ASTERISK-24502 - Build fails when dev-mode, dont optimize and coverage are enabled (Reported by Corey Farrell)
* ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua Colp)
* ASTERISK-26034 - T.38 passthrough problem behind firewall due to early nosignal packet (Reported by George Joseph)
* ASTERISK-26030 - call cut because of double Session-Expires header in re-invite after proxy authentication is required (Reported by George Joseph)

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

http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-11.6-cert14

16Ago/16Off

Rilasciato Asterisk 13.8-cert2

Il giorno 15 agosto 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk Asterisk 13.8-cert2.

Dal post originale:

The release of Certified Asterisk 13.8-cert2 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-26280 - DNS lookups can block channel media paths (Reported by Mark Michelson)
* ASTERISK-26132 - PJSIP: provide transport type with received messages (Reported by Scott Griepentrog)
* 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-26214 - Allow arbitrary time for fax detection to end on a channel (Reported by Richard Mudgett)

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

http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.8-cert2

4Ago/16Off

Rilasciato Asterisk 13.1-cert8

Il giorno 03 agosto 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk Asterisk 13.1-cert8.

Dal post originale:

he release of Certified Asterisk 13.1-cert8 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-26179 - chan_sip: Second T.38 request fails (Reported by Joshua Colp)
* ASTERISK-26089 - Invalid security events during boot using PJSIP Realtime (Reported by Scott Griepentrog)
* ASTERISK-26034 - T.38 passthrough problem behind firewall due to early nosignal packet (Reported by George Joseph)
* ASTERISK-26030 - call cut because of double Session-Expires header in re-invite after proxy authentication is required (Reported by George Joseph)
* ASTERISK-25998 - file: Crash when using nativeformats (Reported by Joshua Colp)
* ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity when calling from Gosub (Reported by Jacques Peacock)
* ASTERISK-25033 - Asterisk 13 (branch head) won't compile without PJSip (Reported by Peter Whisker)

Improvements made in this release:
-----------------------------------
* ASTERISK-26088 - Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett)

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

http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.1-cert8

30Lug/16Off

Presentato Asterisk 14 Beta 1

Dal blog.asterisk.org del 27 luglio 2016.

Dal post originale (http://blogs.asterisk.org/2016/07/27/asterisk-14-beta-1/:

asterisk 14 beta 1 - img1
asterisk 14 beta 1 - img 2