Asterisk Development Team
2015-Apr-01 19:01 UTC
[asterisk-users] Asterisk 11.17.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.17.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.17.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
(Reported by Dwayne Hubbard)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
res_odbc (Reported by ibercom)
* ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
with replaces (Reported by Eelco Brolman)
* ASTERISK-24479 - Enable REF_DEBUG for module references
(Reported by Corey Farrell)
* ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
fully disconnect underlying socket, leading to events being
dropped with no additional information (Reported by Matt Jordan)
* ASTERISK-24772 - ODBC error in realtime sippeers when device
unregisters under MariaDB (Reported by Richard Miller)
* ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
(Reported by Corey Farrell)
* ASTERISK-24799 - [patch] make fails with undefined reference to
SSLv3_client_method (Reported by Alexander Traud)
* ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
for playing back messages stored in IMAP - play_message: No
origtime (Reported by Graham Barnett)
* ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
OSX with 64 bit integers (Reported by Corey Farrell)
* ASTERISK-24796 - Codecs and bucket schema's prevent module
unload (Reported by Corey Farrell)
* ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
(Reported by Ashley Sanders)
* ASTERISK-24797 - bridge_softmix: G.729 codec license held
(Reported by Kevin Harwell)
* ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
thread ID being passed to pthread_kill (Reported by JoshE)
* ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
fail (Reported by Terry Wilson)
* ASTERISK-23214 - chan_sip WARNING message 'We are requesting
SRTP for audio, but they responded without it' is ambiguous and
wrong in some cases (Reported by Rusty Newton)
* ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
error response and BYE are sent to the caller (Reported by
Makoto Dei)
* ASTERISK-18105 - most of asterisk modules are unbuildable in
cygwin environment (Reported by feyfre)
* ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
* ASTERISK-24838 - chan_sip: Locking inversion occurs when
building a peer causes a peer poke during request handling
(Reported by Richard Mudgett)
* ASTERISK-24825 - Caller ID not recognized using
Centrex/Distinctive dialing (Reported by Richard Mudgett)
* ASTERISK-24739 - [patch] - Out of files -- call fails --
numerous files with inodes from under /usr/share/zoneinfo,
mostly posixrules (Reported by Ed Hynan)
* ASTERISK-23390 - NewExten Event with application AGI shows up
before and after AGI runs (Reported by Benjamin Keith Ford)
* ASTERISK-24786 - [patch] - Asterisk terminates when playing a
voicemail stored in LDAP (Reported by Graham Barnett)
* ASTERISK-24808 - res_config_odbc: Improper escaping of
backslashes occurs with MySQL (Reported by Javier Acosta)
* ASTERISK-20850 - [patch]Nested functions aren't portable.
Adapting RAII_VAR to use clang/llvm blocks to get the
same/similar functionality. (Reported by Diederik de Groot)
* ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
by Frank DiGennaro)
* ASTERISK-21038 - Bad command completion of "core set debug
channel" (Reported by Richard Kenner)
* ASTERISK-18708 - func_curl hangs channel under load (Reported by
Dave Cabot)
* ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
Atis Lezdins)
* ASTERISK-24876 - Investigate reference leaks from
tests/channels/local/local_optimize_away (Reported by Corey
Farrell)
* ASTERISK-24817 - init_logger_chain: unreachable code block
(Reported by Corey Farrell)
* ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported by
snuffy)
* ASTERISK-24879 - [patch]Compilation fails due to 64bit time
under OpenBSD (Reported by snuffy)
Improvements made in this release:
-----------------------------------
* ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
Couldn't find mailbox %s in context (Reported by Graham Barnett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
Thank you for your continued support of Asterisk!
Hi list can i ask U does this release solved my problem https://github.com/versatica/JsSIP/issues/311 (already try on a last FREEPBX --same issue) Regards 2015-04-01 22:01 GMT+03:00 Asterisk Development Team < asteriskteam at digium.com>:> The Asterisk Development Team has announced the release of Asterisk > 11.17.0. > This release is available for immediate download at > http://downloads.asterisk.org/pub/telephony/asterisk > > The release of Asterisk 11.17.0 resolves several issues reported by the > community and would have not been possible without your participation. > Thank you! > > The following are the issues resolved in this release: > > New Features made in this release: > ----------------------------------- > * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation > (Reported by Dwayne Hubbard) > > Bugs fixed in this release: > ----------------------------------- > * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in > res_odbc (Reported by ibercom) > * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE > with replaces (Reported by Eelco Brolman) > * ASTERISK-24479 - Enable REF_DEBUG for module references > (Reported by Corey Farrell) > * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to > fully disconnect underlying socket, leading to events being > dropped with no additional information (Reported by Matt Jordan) > * ASTERISK-24772 - ODBC error in realtime sippeers when device > unregisters under MariaDB (Reported by Richard Miller) > * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove > (Reported by Corey Farrell) > * ASTERISK-24799 - [patch] make fails with undefined reference to > SSLv3_client_method (Reported by Alexander Traud) > * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility > for playing back messages stored in IMAP - play_message: No > origtime (Reported by Graham Barnett) > * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc > OSX with 64 bit integers (Reported by Corey Farrell) > * ASTERISK-24796 - Codecs and bucket schema's prevent module > unload (Reported by Corey Farrell) > * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML > (Reported by Ashley Sanders) > * ASTERISK-24797 - bridge_softmix: G.729 codec license held > (Reported by Kevin Harwell) > * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid > thread ID being passed to pthread_kill (Reported by JoshE) > * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime > fail (Reported by Terry Wilson) > * ASTERISK-23214 - chan_sip WARNING message 'We are requesting > SRTP for audio, but they responded without it' is ambiguous and > wrong in some cases (Reported by Rusty Newton) > * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an > error response and BYE are sent to the caller (Reported by > Makoto Dei) > * ASTERISK-18105 - most of asterisk modules are unbuildable in > cygwin environment (Reported by feyfre) > * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell) > * ASTERISK-24838 - chan_sip: Locking inversion occurs when > building a peer causes a peer poke during request handling > (Reported by Richard Mudgett) > * ASTERISK-24825 - Caller ID not recognized using > Centrex/Distinctive dialing (Reported by Richard Mudgett) > * ASTERISK-24739 - [patch] - Out of files -- call fails -- > numerous files with inodes from under /usr/share/zoneinfo, > mostly posixrules (Reported by Ed Hynan) > * ASTERISK-23390 - NewExten Event with application AGI shows up > before and after AGI runs (Reported by Benjamin Keith Ford) > * ASTERISK-24786 - [patch] - Asterisk terminates when playing a > voicemail stored in LDAP (Reported by Graham Barnett) > * ASTERISK-24808 - res_config_odbc: Improper escaping of > backslashes occurs with MySQL (Reported by Javier Acosta) > * ASTERISK-20850 - [patch]Nested functions aren't portable. > Adapting RAII_VAR to use clang/llvm blocks to get the > same/similar functionality. (Reported by Diederik de Groot) > * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported > by Frank DiGennaro) > * ASTERISK-21038 - Bad command completion of "core set debug > channel" (Reported by Richard Kenner) > * ASTERISK-18708 - func_curl hangs channel under load (Reported by > Dave Cabot) > * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by > Atis Lezdins) > * ASTERISK-24876 - Investigate reference leaks from > tests/channels/local/local_optimize_away (Reported by Corey > Farrell) > * ASTERISK-24817 - init_logger_chain: unreachable code block > (Reported by Corey Farrell) > * ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported by > snuffy) > * ASTERISK-24879 - [patch]Compilation fails due to 64bit time > under OpenBSD (Reported by snuffy) > > Improvements made in this release: > ----------------------------------- > * ASTERISK-24790 - Reduce spurious noise in logs from voicemail - > Couldn't find mailbox %s in context (Reported by Graham Barnett) > > For a full list of changes in this release, please see the ChangeLog: > > http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0 > > Thank you for your continued support of Asterisk! > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Best regards Antony ??? (066) 919-75-33 ??? (063) 656-43-40 satskiy.a at gmail.com <mail%3Asatskiy.a at gmail.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150402/9a35cc72/attachment.html>