Bugfix release for 0.4.0, notably for:
* issue with the "oss" plug-in
* less warnings when building with Gtk+ 3
DeforaOS Phone 0.4.0 did bring:
* support for Gtk+ 3 (except for the new "video" plug-in)
* new and updated plug-ins and tools
* additional features and interface updates
* as well as improved documentation (manual pages...)
* and additional improvements under the hood (portability, XDG compliance...)
Also drops the dependency on audio/pulseaudio.
The Asterisk Development Team has announced the release of Asterisk 11.13.0.
The release of Asterisk 11.13.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-24032 - Gentoo compilation emits warning:
"_FORTIFY_SOURCE" redefined (Reported by Kilburn)
* ASTERISK-24225 - Dial option z is broken (Reported by
dimitripietro)
* ASTERISK-24178 - [patch]fromdomainport used even if not set
(Reported by Elazar Broad)
* ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload
warnings and ref leaks (Reported by Walter Doekes)
* ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP
ICE candidates in SDP answer (Reported by Badalian Vyacheslav)
* ASTERISK-24019 - When a Music On Hold stream starts it restarts
at beginning of file. (Reported by Jason Richards)
* ASTERISK-23767 - [patch] Dynamic IAX2 registration stops trying
if ever not able to resolve (Reported by David Herselman)
* ASTERISK-24211 - testsuite: Fix the dial_LS_options test
(Reported by Matt Jordan)
* ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash
Mohod)
* ASTERISK-23577 - res_rtp_asterisk: Crash in
ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by
Jay Jideliov)
* ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10)
concurrent WebRTC (avpg/encryption/icesupport) calls (Reported
by Roman Skvirsky)
* ASTERISK-24301 - Security: Out of call MESSAGE requests
processed via Message channel driver can crash Asterisk
(Reported by Matt Jordan)
Improvements made in this release:
-----------------------------------
* ASTERISK-24171 - [patch] Provide a manpage for the aelparse
utility (Reported by Jeremy Lainé)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.13.0
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 1.8.31.0.
The release of Asterisk 1.8.31.0 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-24032 - Gentoo compilation emits warning:
"_FORTIFY_SOURCE" redefined (Reported by Kilburn)
* ASTERISK-24225 - Dial option z is broken (Reported by
dimitripietro)
* ASTERISK-24178 - [patch]fromdomainport used even if not set
(Reported by Elazar Broad)
* ASTERISK-24019 - When a Music On Hold stream starts it restarts
at beginning of file. (Reported by Jason Richards)
* ASTERISK-24211 - testsuite: Fix the dial_LS_options test
(Reported by Matt Jordan)
* ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash
Mohod)
Improvements made in this release:
-----------------------------------
* ASTERISK-24171 - [patch] Provide a manpage for the aelparse
utility (Reported by Jeremy Lainé)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.31.0
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced security releases for Certified
Asterisk 11.6 and Asterisk 11 and 12. The available security releases are
released as versions 11.6-cert6, 11.12.1, and 12.5.1.
Please note that the release of these versions resolves the following security
vulnerability:
* AST-2014-010: Remote Crash when Handling Out of Call Message in Certain
Dialplan Configurations
Note that the crash described in AST-2014-010 can be worked around through
dialplan configuration. Given the likelihood of the issue, an advisory was
deemed to be warranted.
For more information about the details of these vulnerabilities, please read
security advisories AST-2014-009 and AST-2014-010, which were released at the
same time as this announcement.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.12.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2014-010.pdf
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 1.8.30.0.
The release of Asterisk 1.8.30.0 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-23911 - URIENCODE/URIDECODE: WARNING about passing an
empty string is a bit over zealous (Reported by Matt Jordan)
* ASTERISK-23814 - No call started after peer dialed (Reported by
Igor Goncharovsky)
* ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy
should not call sip_destroy (Reported by Corey Farrell)
* ASTERISK-23818 - PBX_Lua: after asterisk startup module is
loaded, but dialplan not available (Reported by Dennis Guse)
* ASTERISK-18345 - [patch] sips connection dropped by asterisk
with a large INVITE (Reported by Stephane Chazelas)
* ASTERISK-23508 - Memory Corruption in
__ast_string_field_ptr_build_va (Reported by Arnd Schmitter)
Improvements made in this release:
-----------------------------------
* ASTERISK-21178 - Improve documentation for manager command
Getvar, Setvar (Reported by Rusty Newton)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.30.0
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 11.12.0.
The release of Asterisk 11.12.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-23911 - URIENCODE/URIDECODE: WARNING about passing an
empty string is a bit over zealous (Reported by Matt Jordan)
* ASTERISK-23985 - PresenceState Action response does not contain
ActionID; duplicates Message Header (Reported by Matt Jordan)
* ASTERISK-23814 - No call started after peer dialed (Reported by
Igor Goncharovsky)
* ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy
should not call sip_destroy (Reported by Corey Farrell)
* ASTERISK-23818 - PBX_Lua: after asterisk startup module is
loaded, but dialplan not available (Reported by Dennis Guse)
* ASTERISK-18345 - [patch] sips connection dropped by asterisk
with a large INVITE (Reported by Stephane Chazelas)
* ASTERISK-23508 - Memory Corruption in
__ast_string_field_ptr_build_va (Reported by Arnd Schmitter)
Improvements made in this release:
-----------------------------------
* ASTERISK-21178 - Improve documentation for manager command
Getvar, Setvar (Reported by Rusty Newton)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.12.0
Thank you for your continued support of Asterisk!
pkgsrc change: MAKE_JOBS_SAFE=NO from joerg@
The Asterisk Development Team has announced the release of Asterisk 11.11.0.
The release of Asterisk 11.11.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-22551 - Session timer : UAS (Asterisk) starts counting
at Invite, UAC starts counting at 200 OK. (Reported by i2045)
* ASTERISK-23792 - Mutex left locked in chan_unistim.c (Reported
by Peter Whisker)
* ASTERISK-23582 - [patch]Inconsistent column length in *odbc
(Reported by Walter Doekes)
* ASTERISK-23803 - AMI action UpdateConfig EmptyCat clears all
categories but the requested one (Reported by zvision)
* ASTERISK-23035 - ConfBridge with name longer than max (32 chars)
results in several bridges with same conf_name (Reported by
Iñaki Cívico)
* ASTERISK-23824 - ConfBridge: Users cannot be muted via CLI or
AMI when waiting to enter a conference (Reported by Matt Jordan)
* ASTERISK-23683 - #includes - wildcard character in a path more
than one directory deep - results in no config parsing on module
reload (Reported by tootai)
* ASTERISK-23827 - autoservice thread doesn't exit at shutdown
(Reported by Corey Farrell)
* ASTERISK-23609 - Security: AMI action MixMonitor allows
arbitrary programs to be run (Reported by Corey Farrell)
* ASTERISK-23673 - Security: DOS by consuming the number of
allowed HTTP connections. (Reported by Richard Mudgett)
* ASTERISK-23246 - DEBUG messages in sdp_crypto.c display despite
a DEBUG level of zero (Reported by Rusty Newton)
* ASTERISK-23766 - [patch] Specify timeout for database write in
SQLite (Reported by Igor Goncharovsky)
* ASTERISK-23844 - Load of pbx_lua fails on sample extensions.lua
with Lua 5.2 or greater due to addition of goto statement
(Reported by Rusty Newton)
* ASTERISK-23818 - PBX_Lua: after asterisk startup module is
loaded, but dialplan not available (Reported by Dennis Guse)
* ASTERISK-23834 - res_rtp_asterisk debug message gives wrong
length if ICE (Reported by Richard Kenner)
* ASTERISK-23790 - [patch] - SIP From headers longer than 256
characters result in dropped call and 'No closing bracket'
warnings. (Reported by uniken1)
* ASTERISK-23917 - res_http_websocket: Delay in client processing
large streams of data causes disconnect and stuck socket
(Reported by Matt Jordan)
* ASTERISK-23908 - [patch]When using FEC error correction,
asterisk tries considers negative sequence numbers as missing
(Reported by Torrey Searle)
* ASTERISK-23921 - refcounter.py uses excessive ram for large refs
files (Reported by Corey Farrell)
* ASTERISK-23948 - REF_DEBUG fails to record ao2_ref against
objects that were already freed (Reported by Corey Farrell)
* ASTERISK-23916 - [patch]SIP/SDP fmtp line may include whitespace
between attributes (Reported by Alexander Traud)
* ASTERISK-23984 - Infinite loop possible in ast_careful_fwrite()
(Reported by Steve Davies)
* ASTERISK-23897 - [patch]Change in SETUP ACK handling (checking
PI) in revision 413765 breaks working environments (Reported by
Pavel Troller)
Improvements made in this release:
-----------------------------------
* ASTERISK-23492 - Add option to safe_asterisk to disable
backgrounding (Reported by Walter Doekes)
* ASTERISK-22961 - [patch] DTLS-SRTP not working with SHA-256
(Reported by Jay Jideliov)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.11.0
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 1.8.29.0.
The release of Asterisk 1.8.29.0 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-22551 - Session timer : UAS (Asterisk) starts counting
at Invite, UAC starts counting at 200 OK. (Reported by i2045)
* ASTERISK-23582 - [patch]Inconsistent column length in *odbc
(Reported by Walter Doekes)
* ASTERISK-23803 - AMI action UpdateConfig EmptyCat clears all
categories but the requested one (Reported by zvision)
* ASTERISK-23035 - ConfBridge with name longer than max (32 chars)
results in several bridges with same conf_name (Reported by
Iñaki Cívico)
* ASTERISK-23683 - #includes - wildcard character in a path more
than one directory deep - results in no config parsing on module
reload (Reported by tootai)
* ASTERISK-23827 - autoservice thread doesn't exit at shutdown
(Reported by Corey Farrell)
* ASTERISK-23814 - No call started after peer dialed (Reported by
Igor Goncharovsky)
* ASTERISK-23673 - Security: DOS by consuming the number of
allowed HTTP connections. (Reported by Richard Mudgett)
* ASTERISK-23246 - DEBUG messages in sdp_crypto.c display despite
a DEBUG level of zero (Reported by Rusty Newton)
* ASTERISK-23766 - [patch] Specify timeout for database write in
SQLite (Reported by Igor Goncharovsky)
* ASTERISK-23818 - PBX_Lua: after asterisk startup module is
loaded, but dialplan not available (Reported by Dennis Guse)
* ASTERISK-23667 - features.conf.sample is unclear as to which
options can or cannot be set in the general section (Reported by
David Brillert)
* ASTERISK-23790 - [patch] - SIP From headers longer than 256
characters result in dropped call and 'No closing bracket'
warnings. (Reported by uniken1)
* ASTERISK-23908 - [patch]When using FEC error correction,
asterisk tries considers negative sequence numbers as missing
(Reported by Torrey Searle)
* ASTERISK-23921 - refcounter.py uses excessive ram for large refs
files (Reported by Corey Farrell)
* ASTERISK-23948 - REF_DEBUG fails to record ao2_ref against
objects that were already freed (Reported by Corey Farrell)
* ASTERISK-23984 - Infinite loop possible in ast_careful_fwrite()
(Reported by Steve Davies)
* ASTERISK-23897 - [patch]Change in SETUP ACK handling (checking
PI) in revision 413765 breaks working environments (Reported by
Pavel Troller)
Improvements made in this release:
-----------------------------------
* ASTERISK-23564 - [patch]TLS/SRTP status of channel not currently
available in a CLI command (Reported by Patrick Laimbock)
* ASTERISK-23492 - Add option to safe_asterisk to disable
backgrounding (Reported by Walter Doekes)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.29.0
Thank you for your continued support of Asterisk!
* Depend on x11/c++-gtk-utils, instead of x11/gtkmm
Changelog:
Version 3.2.13 (11th April 2014)
--------------
Fix API breakage in GTK+-3.12 (the buttons of GtkDialog objects
have been made internal instead of non-internal children of the
action area box) (dialog.h, dialog.cpp).
Put icon in efax and efax-gtk about dialogs (dialogs.cpp).
Replace AC_CONFIG_HEADER macro with AC_CONFIG_HEADERS when
configuring (Samuli Suominen) (configure.ac).
Update build system to automake-1.13.3 (config.guess, config.sub,
depcomp, INSTALL).
Update desktop file (Samuli Suominen) (efax-gtk.desktop).
Version 3.2.12 (1st June 2013)
--------------
Force GType initialisation of GtkEntry for GtkSettings
(mainwindow.cpp).
Correct entry sizing in settings dialog (settings.cpp).
Improve tray icon sizing (tray_icon.cpp).
Permit the program to build against c++-gtk-utils-2.2 (this
requires increasing the c++-gtk-utils-1.2 dependency to 1.2.13,
and increasing the c++-gtk-utils-2.0 dependency to 2.0.1)
(acinclude.m4, README; mainwindow.h, mainwindow.cpp).
Cause bootstrap.sh to build translation files (bootstrap.sh).
Update build system to automake-1.13.1 (configure.ac,
config.guess, config.sub, depcomp, INSTALL, install-sh, missing;
src/Makefile.am; efax/Makefile.am).
Version 3.2.11 (1st January 2013)
--------------
Workaround for a bug in GtkFileChooserDialog in later versions of
gtk+-2.24 (dialogs.cpp).
Change efax-gtk.desktop to meet
http://specifications.freedesktop.org/menu-spec/menu-spec-latest.html
recommendations (efax-gtk.desktop).
Add French translation (Charlie Ledocq) (po/fr.po, LINGUAS).
Version 3.2.10 (21st October 2012)
--------------
Update build system to automake-1.12.1 and autoconf-2.69.
Suppress gtk+-3 deprecation warnings (acinclude.m4).
Set locale even if NLS not set (main.cpp).
Deal better with GtkMessageDialog format string (dialog.cpp).
Use automake silent rules (configure.ac).
Correct icon entry in efax-gtk.desktop file (efax-gtk.desktop).
Simplify file chooser selection code (dialogs.cpp).
Call atexit() instead of glib's now deprecated g_atexit() (the use
of atexit() in this program is entirely safe) (main.cpp).
Remove unnecessary pointer value check in present_prog()
(main.cpp).
Add Spanish translation (Antonio Trujillo) (po/LINGUAS, po/es.po,
po/efax-gtk.pot; mainwindow.cpp).
Update copyright notices (COPYING, README; addressbook.h,
addressbook.cpp, dialogs.h, dialogs.cpp, efax_controller.h,
efax_controller.cpp, fax_list.h, fax_list.cpp, fax_list_manager.h,
fax_list_manager.cpp, file_list.h, file_list.cp, gpl.h,
helpfile.h, helpfile.cpp, logger.h, logger.cpp, main.cpp,
mainwindow.h, mainwindow.cpp, prog_defs.h, redial_queue.h,
redial_queue.cpp, settings.h, settings.cpp, settings_help.h,
settings_help.cpp, socket_list.h, socket_list.cpp,
socket_notify.h, socket_notify.cpp, socket_server.h,
socket_server.cpp, tray_icon.h, tray_icon.cpp;
utils/cairo_handle.h, utils/icon_info_handle.h,
utils/mono_tiff_print_manager.h,
utils/mono_tiff_print_manager.cpp,
utils/pango_layout_iter_handle.h, utils/selected_rows_handle.h,
utils/selected_rows_handle.cpp, utils/sem_sync.h,
utils/tiff_handle.h, utils/toolbar_append_widget.h,
utils/toolbar_append_widget.cpp, utils/tree_path_handle.h,
utils/tree_row_reference_handle.h, utils/utf8_utils.h,
utils/utf8_utils.cpp;
efax-gtk-faxfilter/efax-gtk-socket-client.cpp.
Update documentation (README).
Update configuration scripts (acinclude.m4).
Version 3.2.9 (21st December 2011)
-------------
Have a hard dependency on c++-gtk-utils-1.2 >= 1.2.7 or
c++-gtk-utils-2.0 >= 2.0.0-rc1, so that C++0x/11 can be more
easily supported, and maintainability is improved (README,
acinclude.m4, configure.ac, src/Makefile.am,
src/utils/Makefile.am, po/POTFILES.in; delete src/internal
directory and its contents.)
Make the settings help dialogs of a reasonable size with GTK+3
(settings_help.cpp).
Permit IPv6 addresses to be specified with a wildcard and only
one, or no, ':' character (socket_server.cpp).
Explicitly set shadow type of fax input frame (mainwindow.cpp).
Change library linking order (src/Makefile.am and
efax/Makefile.am).
Remove unnecessary configure checks (acinclude.m4, configure.ac
and src/Makefile.am)
Upgrade gettext to version 0.18 and include m4 macros
(po/Makefile.in.in, po/Rules-quot, m4 directory, Makefile.am).
Add proper header checks at configuration time (configure.ac).
Version 3.2.8 (30th March 2011)
-------------
Permit IPv6 addresses in domain name form as well as in numeric
format (socket_server.h and socket_server.cpp).
Improve error checking and so suppress gcc-4.6 warning
(mainwindow.cpp; utils/mono_tiff_print_manager.cpp).
Upgrade internal c++-gtk-utils version.
Version 3.2.7.1 (16th March 2011)
---------------
Correct style change handling (mainwindow.h, mainwindow.cpp).
Update comments in efax-gtkrc on "SOCK_OTHER_ADDRESSES:" for IPv6
(efax-gtkrc).
Version 3.2.7 (14th March 2011)
-------------
Change default gtk target to gtk+3, and permit
--with-gtk-version=gtk2 and --with=gtk-version=gtk3 as well as
--with-gtk-version=gtk+2 and --with=gtk-version=gtk+3
(acinclude.m4, README).
Provide option for server to accept IPv6 connections (efax-gtkrc;
prog_defs.h, main.cpp, mainwindow.cpp, settings.h, settings.cpp,
settings_help.h, settings_help.cpp, socket_server.h and
socket_server.cpp).
Use GtkStyleContext for a GTK+3 compile (acinclude.m4;
addressbook.cpp, dialogs.h, dialogs.cpp, fax_list.cpp,
fax_list-manager.cpp, helpfile.cpp, logger.cpp, main.cpp,
mainwindow.h and mainwindow.cpp).
Correct non-sh-ism in configuration files (acinclude.m4).
Include efax-gtk.png icon in rpm spec file (efax-gtk.spec.in).
Minor adjustments to MonoTiffPrintManager implementation
(utils/mono_tiff_print_manager.h and
utils/mono_tiff_print_manager.cpp).
Provide some explicit casts for std::pair constructor required by
C++0x (dialogs.cpp, fax_list.cpp, file_list.cpp, socket_list.cpp).
Provide compile option for c++-gtk-utils-2.0 (acinclude.m4,
README; efax_controller.cpp, fax_list.cpp, gpl.h, logger.cpp,
socket_server.h and socket_server.cpp).
Surpress warning about std::auto_ptr being deprecated when
compiling under C++0x (acinclude.m4).
Upgrade internal c++-gtk-utils version to 1.2.12 (and also adjust
the configuration files for that (acinclude.m4, configure.ac;
src/internal/c++-gtk-utils/Makefile.am)).
Version 3.2.6 (13th November 2010)
-------------
Do not require dbus-glib where glib >= 2.26 is installed
(acinclude.4, configure.ac, src/Makefile.am,
src/internal/c++-gtk-utils/Makefile.am with upgraded
c++-gtk-utils).
Fixes for gtk+-2.91 (acinclude.m4, addressbook.cpp, dialogs.cpp,
fax_list.cpp, fax_list_manager.cpp, helpfile.cpp, logger.cpp,
main.cpp, mainwindow.h, mainwindow.cpp, settings.cpp and
socket_notify.cpp).
Improve widget sizing for widgets displaying text
(efax_controller.h, mainwindow.h and mainwindow.cpp).
Fix compilation error with gtk+-2.12 (mainwindow.cpp).
Remove anachronistic comment (utils/mono_tiff_print_manager.cpp).
Upgrade internal c++-gtk-utils version to 1.2.7.
Version 3.2.5 (13th October 2010)
-------------
Fix segfault when printing faxes with cairo-1.10
(utils/mono_tiff_print_manager.h and
utils/mono_tiff_print_manager.cpp).
Use cairo rather than the GDK drawing functions to draw the
indicator of whether there are print jobs from the socket to be
faxed (mainwindow.h, mainwindow.cpp and utils/cairo_handle.h).
Use gtk_tree_view_convert_bin_window_to_widget_coords() instead of
gtk_widget_get_pointer() in order to obtain the pointer position
in widget co-ordinates for tree view motion notify events
(fax_list_manager.cpp).
Modify argument handling for efax message functions to avoid an
invalid double call to vfprintf() on the same va_list value (this
bug is triggered on some systems when using the 'fax' script but
does not directly affect efax-gtk) (efax/efaxmsg.c, efax/PATCHES).
Upgrade internal c++-gtk-utils version to 1.2.6.
Version 3.2.4 (2nd August 2010)
-------------
Fix uncaught exception where a file to be faxed is not in valid
postscript/PDF format (efax_controller.cpp).
Include the former gnome stock_send-fax icon as the standard icon
for efax-gtk (Makefile.am, efax-gtk.desktop, efax-gtk.png,
AUTHORS; main.cpp).
Use XkbBell() rather than XBell() where available (acinclude.m4,
main.cpp).
Include pkg-config test for x11.pc, if available (acinclude.m4 and
src/Makefile.am).
Further build fixes for Debian Hurd (src/efax_controller.cpp and
src/fax_list.cpp; efax/efaxmsg.c).
Update Hungarian translation (László Csordás) (hu.po and
mainwindow.cpp).
Remove redundant anonymous namespace for callbacks with C linkage
(addressbook.h, addressbook.cpp, dialogs.h, dialogs.cpp,
fax_list.h, fax_list.cpp, fax_list_manager.h,
fax_list_manager.cpp, file_list.h, file_list.cpp, helpfile.h,
helpfile.cpp, logger.h, logger.cpp, mainwindow.h, mainwindow.cpp,
redial_queue.h, redial_queue.cpp, settings.h, settings.cpp,
settings_help.h, settings_help.cpp, socket_list.h,
socket_list.cpp, socket_notify.h, socket_notify.cpp, tray_icon.h,
tray_icon.cpp, utils/mono_tiff_print_manager.h,
utils/mono_tiff_print_manager.cpp, utils/selected_rows_handle.h,
utils/selected_rows_handle.cpp).
Correct linkage specification of present_prog() (main.cpp).
Use Cgu::start_timeout_seconds() rather than Cgu::start_timeout()
where available (fax_list_manager.cpp, logger.cpp).
Remove redundant comments (prog_defs.h,
utils/toolbar_append_widget.cpp).
Upgrade internal c++-gtk-utils version to 1.2.4.
numerous general bugs. The vulnerabilities fixed are: AST-2014-001,
AST-2014-002, and AST-2014-007.
-----
The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The
available security releases are released as versions 1.8.15-cert7,
11.6-cert4, 1.8.28.2, 11.10.2, and 12.3.2.
These releases resolve security vulnerabilities that were previously
fixed in 1.8.15-cert6, 11.6-cert3, 1.8.28.1, 11.10.1, and 12.3.1.
Unfortunately, the fix for AST-2014-007 inadvertently introduced
a regression in Asterisk's TCP and TLS handling that prevented
Asterisk from sending data over these transports. This regression
and the security vulnerabilities have been fixed in the versions
specified in this release announcement.
The security patches for AST-2014-007 have been updated with the
fix for the regression, and are available at
http://downloads.asterisk.org/pub/security
Please note that the release of these versions resolves the following security
vulnerabilities:
* AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP
Connections
For more information about the details of these vulnerabilities,
please read security advisories AST-2014-005, AST-2014-006,
AST-2014-007, and AST-2014-008, which were released with the previous
versions that addressed these vulnerabilities.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.28.2
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2014-007.pdf
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The
available security releases are released as versions 1.8.15-cert6,
11.6-cert3, 1.8.28.1, 11.10.1, and 12.3.1.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The release of these versions resolves the following issue:
* AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP
Connections
Establishing a TCP or TLS connection to the configured HTTP or
HTTPS port respectively in http.conf and then not sending or
completing a HTTP request will tie up a HTTP session. By doing
this repeatedly until the maximum number of open HTTP sessions
is reached, legitimate requests are blocked.
These issues and their resolutions are described in the security advisories.
For more information about the details of these vulnerabilities,
please read security advisories AST-2014-005, AST-2014-006,
AST-2014-007, and AST-2014-008, which were released at the same
time as this announcement.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.28.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2014-007.pdf
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 1.8.28.0.
The release of Asterisk 1.8.28.0 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-23547 - [patch] app_queue removing callers from queue
when reloading (Reported by Italo Rossi)
* ASTERISK-22846 - testsuite: masquerade super test fails on all
branches (still) (Reported by Matt Jordan)
* ASTERISK-23546 - CB_ADD_LEN does not do what you'd think
(Reported by Walter Doekes)
* ASTERISK-23620 - Code path in app_stack fails to unlock list
(Reported by Bradley Watkins)
* ASTERISK-18331 - app_sms failure (Reported by David Woodhouse)
* ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by
Krzysztof Chmielewski)
* ASTERISK-23707 - Realtime Contacts: Apparent mismatch between
PGSQL database state and Asterisk state (Reported by Mark
Michelson)
* ASTERISK-23665 - Wrong mime type for codec H263-1998 (h263+)
(Reported by Guillaume Maudoux)
* ASTERISK-22977 - chan_sip+CEL: missing ANSWER and PICKUP event
for INVITE/w/replaces pickup (Reported by Walter Doekes)
* ASTERISK-23709 - Regression in Dahdi/Analog/waitfordialtone
(Reported by Steve Davies)
* ASTERISK-23650 - Intermittent segfault in string functions
(Reported by Roel van Meer)
Improvements made in this release:
-----------------------------------
* ASTERISK-23754 - [patch] Use var/lib directory for log file
configured in asterisk.conf (Reported by Igor Goncharovsky)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.28.0
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 1.8.27.0.
The release of Asterisk 1.8.27.0 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-22790 - check_modem_rate() may return incorrect rate
for V.27 (Reported by Paolo Compagnini)
* ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in
sip.conf.sample (Reported by Eugene)
* ASTERISK-23028 - [patch] Asterisk man pages contains unquoted
minus signs (Reported by Jeremy Lainé)
* ASTERISK-23046 - Custom CDR fields set during a GoSUB called
from app_queue are not inserted (Reported by Denis Pantsyrev)
* ASTERISK-23027 - [patch] Spelling typo "transfered" instead of
"transferred" (Reported by Jeremy Lainé)
* ASTERISK-23008 - Local channels loose CALLERID name when DAHDI
channel connects (Reported by Michael Cargile)
* ASTERISK-23100 - [patch] In chan_mgcp the ident in transmitted
request and request queue may differ - fix for locking (Reported
by adomjan)
* ASTERISK-22988 - [patch]T38 , SIP 488 after Rejecting image
media offer due to invalid or unsupported syntax (Reported by
adomjan)
* ASTERISK-22861 - [patch]Specifying a null time as parameter to
GotoIfTime or ExecIfTime causes segmentation fault (Reported by
Sebastian Murray-Roberts)
* ASTERISK-17837 - extconfig.conf - Maximum Include level (1)
exceeded (Reported by pz)
* ASTERISK-22662 - Documentation fix? - queues.conf says
persistentmembers defaults to yes, it appears to lie (Reported
by Rusty Newton)
* ASTERISK-23134 - [patch] res_rtp_asterisk port selection cannot
handle selinux port restrictions (Reported by Corey Farrell)
* ASTERISK-23220 - STACK_PEEK function with no arguments causes
crash/core dump (Reported by James Sharp)
* ASTERISK-19773 - Asterisk crash on issuing Asterisk-CLI 'reload'
command multiple times on cli_aliases (Reported by Joel Vandal)
* ASTERISK-22757 - segfault in res_clialiases.so on reload when
mapping "module reload" command (Reported by Gareth Blades)
* ASTERISK-17727 - [patch] TLS doesn't get all certificate chain
(Reported by LN)
* ASTERISK-23178 - devicestate.h: device state setting functions
are documented with the wrong return values (Reported by
Jonathan Rose)
* ASTERISK-23297 - Asterisk 12, pbx_config.so segfaults if
res_parking.so is not loaded, or if res_parking.conf has no
configuration (Reported by CJ Oster)
* ASTERISK-23069 - Custom CDR variable not recorded when set in
macro called from app_queue (Reported by Bryan Anderson)
* ASTERISK-19499 - ConfBridge MOH is not working for transferee
after attended transfer (Reported by Timo Teräs)
* ASTERISK-23261 - [patch]Output mixup in
${CHANNEL(rtpqos,audio,all)} (Reported by rsw686)
* ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR
variables for subsequent records (Reported by zvision)
* ASTERISK-23141 - Asterisk crashes on Dial(), in
pbx_find_extension at pbx.c (Reported by Maxim)
* ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
to minrate=2400, then res_fax refuse to load (Reported by David
Brillert)
* ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set
- probably introduced in 11.7.0 (Reported by OK)
* ASTERISK-23323 - [patch]chan_sip: missing p->owner checks in
handle_response_invite (Reported by Walter Doekes)
* ASTERISK-23382 - [patch]Build System: make -qp can corrupt
menuselect-tree and related files (Reported by Corey Farrell)
* ASTERISK-23406 - [patch]Fix typo in "sip show peer" (Reported by
ibercom)
* ASTERISK-23310 - bridged channel crashes in bridge_p2p_rtp_write
(Reported by Jeremy Lainé)
* ASTERISK-23104 - Specifying the SetVar AMI without a Channel
cause Asterisk to crash (Reported by Joel Vandal)
* ASTERISK-23383 - Wrong sense test on stat return code causes
unchanged config check to break with include files. (Reported by
David Woolley)
* ASTERISK-17523 - Qualify for static realtime peers does not work
(Reported by Maciej Krajewski)
* ASTERISK-21406 - [patch] chan_sip deadlock on monlock between
unload_module and do_monitor (Reported by Corey Farrell)
* ASTERISK-23373 - [patch]Security: Open FD exhaustion with
chan_sip Session-Timers (Reported by Corey Farrell)
* ASTERISK-23340 - Security Vulnerability: stack allocation of
cookie headers in loop allows for unauthenticated remote denial
of service attack (Reported by Matt Jordan)
* ASTERISK-23488 - Logic error in callerid checksum processing
(Reported by Russ Meyerriecks)
* ASTERISK-20841 - fromdomain not honored on outbound INVITE
request (Reported by Kelly Goedert)
* ASTERISK-22079 - Segfault: INTERNAL_OBJ (user_data=0x6374652f)
at astobj2.c:120 (Reported by Jamuel Starkey)
* ASTERISK-23509 - [patch]SayNumber for Polish language tries to
play empty files for numbers divisible by 100 (Reported by
zvision)
* ASTERISK-23391 - Audit dialplan function usage of channel
variable (Reported by Corey Farrell)
* ASTERISK-23548 - POST to ARI sometimes returns no body on
success (Reported by Scott Griepentrog)
Improvements made in this release:
-----------------------------------
* ASTERISK-22980 - [patch]Allow building cdr_radius and cel_radius
against libfreeradius-client (Reported by Jeremy Lainé)
* ASTERISK-22661 - Unable to exit ChanSpy if spied channel does
not have a call in progress (Reported by Chris Hillman)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.27.0
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The
available security releases are released as versions 1.8.15-cert5,
11.6-cert2, 1.8.26.1, 11.8.1, and 12.1.1.
The release of these versions resolve the following issues:
* AST-2014-001: Stack overflow in HTTP processing of Cookie headers.
Sending a HTTP request that is handled by Asterisk with a large number of
Cookie headers could overflow the stack.
Another vulnerability along similar lines is any HTTP request with a
ridiculous number of headers in the request could exhaust system memory.
* AST-2014-002: chan_sip: Exit early on bad session timers request
This change allows chan_sip to avoid creation of the channel and
consumption of associated file descriptors altogether if the inbound
request is going to be rejected anyway.
These issues and their resolutions are described in the security advisories.
For more information about the details of these vulnerabilities,
please read security advisories AST-2014-001, AST-2014-002,
AST-2014-003, and AST-2014-004, which were released at the same
time as this announcement.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.26.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2014-001.pdf
* http://downloads.asterisk.org/pub/security/AST-2014-002.pdf
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 1.8.26.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 1.8.26.0 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-22544 - Italian prompt vm-options has advertisement in
it (Reported by Rusty Newton)
* ASTERISK-12117 - chan_sip creates a new local tag (from-tag) for
every register message (Reported by Pawel Pierscionek)
* ASTERISK-20862 - Asterisk min and max member penalties not
honored when set with 0 (Reported by Schmooze Com)
* ASTERISK-22746 - [patch]Crash in chan_dahdi during caller id
read (Reported by Michael Walton)
* ASTERISK-22788 - [patch] main/translate.c: access to variable f
after free in ast_translate() (Reported by Corey Farrell)
* ASTERISK-21242 - Segfault when T.38 re-invite retransmission
receives 200 OK (Reported by Ashley Winters)
* ASTERISK-22590 - BufferOverflow in unpacksms16() when receiving
16 bit multipart SMS with app_sms (Reported by Jan Juergens)
* ASTERISK-22905 - Prevent Asterisk functions that are 'dangerous'
from being executed from external interfaces (Reported by Matt
Jordan)
* ASTERISK-23021 - Typos in code : "avaliable" instead of
"available" (Reported by Jeremy Lainé)
* ASTERISK-22970 - [patch]Documentation fix for QUOTE() (Reported
by Gareth Palmer)
* ASTERISK-22856 - [patch]SayUnixTime in polish reads minutes
instead of seconds (Reported by Robert Mordec)
* ASTERISK-22854 - [patch] - Deadlock between cel_pgsql unload and
core_event_dispatcher taskprocessor thread (Reported by Etienne
Lessard)
* ASTERISK-22910 - [patch] - REPLACE() calls strcpy on overlapping
memory when <replace-char> is empty (Reported by Gareth Palmer)
* ASTERISK-22871 - cel_pgsql module not loading after "reload" or
"reload cel_pgsql.so" command (Reported by Matteo)
* ASTERISK-23084 - [patch]rasterisk needlessly prints the
AST-2013-007 warning (Reported by Tzafrir Cohen)
* ASTERISK-17138 - [patch] Asterisk not re-registering after it
receives "Forbidden - wrong password on authentication"
(Reported by Rudi)
* ASTERISK-23011 - [patch]configure.ac and pbx_lua don't support
lua 5.2 (Reported by George Joseph)
* ASTERISK-22834 - Parking by blind transfer when lot full orphans
channels (Reported by rsw686)
* ASTERISK-23047 - Orphaned (stuck) channel occurs during a failed
SIP transfer to parking space (Reported by Tommy Thompson)
* ASTERISK-22946 - Local From tag regression with sipgate.de
(Reported by Stephan Eisvogel)
* ASTERISK-23010 - No BYE message sent when sip INVITE is received
(Reported by Ryan Tilton)
Improvements made in this release:
-----------------------------------
* ASTERISK-22659 - Make a new core and extra sounds release
(Reported by Rusty Newton)
* ASTERISK-22918 - dahdi show channels slices PRI channel dnid on
output (Reported by outtolunc)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.26.0
Thank you for your continued support of Asterisk!
with general bug fixes. The security issues fixed are: AST-2014-001,
AST-2014-002, AST-2014-006, and AST-2014-007.
-----
The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The
available security releases are released as versions 1.8.15-cert7,
11.6-cert4, 1.8.28.2, 11.10.2, and 12.3.2.
These releases resolve security vulnerabilities that were previously
fixed in 1.8.15-cert6, 11.6-cert3, 1.8.28.1, 11.10.1, and 12.3.1.
Unfortunately, the fix for AST-2014-007 inadvertently introduced
a regression in Asterisk's TCP and TLS handling that prevented
Asterisk from sending data over these transports. This regression
and the security vulnerabilities have been fixed in the versions
specified in this release announcement.
Please note that the release of these versions resolves the following security
vulnerabilities:
* AST-2014-006: Permission Escalation via Asterisk Manager User Unauthorized
Shell Access
* AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP
Connections
For more information about the details of these vulnerabilities,
please read security advisories AST-2014-005, AST-2014-006,
AST-2014-007, and AST-2014-008, which were released with the previous
versions that addressed these vulnerabilities.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.10.2
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2014-006.pdf
* http://downloads.asterisk.org/pub/security/AST-2014-007.pdf
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The
available security releases are released as versions 1.8.15-cert6,
11.6-cert3, 1.8.28.1, 11.10.1, and 12.3.1.
The release of these versions resolves the following issue:
* AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP
Connections
Establishing a TCP or TLS connection to the configured HTTP or HTTPS port
respectively in http.conf and then not sending or completing a HTTP request
will tie up a HTTP session. By doing this repeatedly until the maximum number
of open HTTP sessions is reached, legitimate requests are blocked.
Additionally, the release of 11.6-cert3, 11.10.1, and 12.3.1 resolves the
following issue:
* AST-2014-006: Permission Escalation via Asterisk Manager User Unauthorized
Shell Access
Manager users can execute arbitrary shell commands with the MixMonitor manager
action. Asterisk does not require system class authorization for a manager
user to use the MixMonitor action, so any manager user who is permitted to use
manager commands can potentially execute shell commands as the user executing
the Asterisk process.
These issues and their resolutions are described in the security advisories.
For more information about the details of these vulnerabilities,
please read security advisories AST-2014-005, AST-2014-006,
AST-2014-007, and AST-2014-008, which were released at the same
time as this announcement.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.10.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2014-006.pdf
* http://downloads.asterisk.org/pub/security/AST-2014-007.pdf
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 11.10.0.
The release of Asterisk 11.10.0 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-23547 - [patch] app_queue removing callers from queue
when reloading (Reported by Italo Rossi)
* ASTERISK-23559 - app_voicemail fails to load after fix to
dialplan functions (Reported by Corey Farrell)
* ASTERISK-22846 - testsuite: masquerade super test fails on all
branches (still) (Reported by Matt Jordan)
* ASTERISK-23545 - Confbridge talker detection settings
configuration load bug (Reported by John Knott)
* ASTERISK-23546 - CB_ADD_LEN does not do what you'd think
(Reported by Walter Doekes)
* ASTERISK-23620 - Code path in app_stack fails to unlock list
(Reported by Bradley Watkins)
* ASTERISK-23616 - Big memory leak in logger.c (Reported by
ibercom)
* ASTERISK-23576 - Build failure on SmartOS / Illumos / SunOS
(Reported by Sebastian Wiedenroth)
* ASTERISK-23550 - Newer sound sets don't show up in menuselect
(Reported by Rusty Newton)
* ASTERISK-18331 - app_sms failure (Reported by David Woodhouse)
* ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by
Krzysztof Chmielewski)
* ASTERISK-23605 - res_http_websocket: Race condition in shutting
down websocket causes crash (Reported by Matt Jordan)
* ASTERISK-23707 - Realtime Contacts: Apparent mismatch between
PGSQL database state and Asterisk state (Reported by Mark
Michelson)
* ASTERISK-23381 - [patch]ChanSpy- Barge only works on the initial
'spy', if the spied-on channel makes a new call, unable to
barge. (Reported by Robert Moss)
* ASTERISK-23665 - Wrong mime type for codec H263-1998 (h263+)
(Reported by Guillaume Maudoux)
* ASTERISK-23664 - Incorrect H264 specification in SDP. (Reported
by Guillaume Maudoux)
* ASTERISK-22977 - chan_sip+CEL: missing ANSWER and PICKUP event
for INVITE/w/replaces pickup (Reported by Walter Doekes)
* ASTERISK-23709 - Regression in Dahdi/Analog/waitfordialtone
(Reported by Steve Davies)
Improvements made in this release:
-----------------------------------
* ASTERISK-23649 - [patch]Support for DTLS retransmission
(Reported by NITESH BANSAL)
* ASTERISK-23564 - [patch]TLS/SRTP status of channel not currently
available in a CLI command (Reported by Patrick Laimbock)
* ASTERISK-23754 - [patch] Use var/lib directory for log file
configured in asterisk.conf (Reported by Igor Goncharovsky)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.10.0
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 11.9.0.
The release of Asterisk 11.9.0 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-22790 - check_modem_rate() may return incorrect rate
for V.27 (Reported by Paolo Compagnini)
* ASTERISK-23034 - [patch] manager Originate doesn't abort on
failed format_cap allocation (Reported by Corey Farrell)
* ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in
sip.conf.sample (Reported by Eugene)
* ASTERISK-23028 - [patch] Asterisk man pages contains unquoted
minus signs (Reported by Jeremy Lainé)
* ASTERISK-23046 - Custom CDR fields set during a GoSUB called
from app_queue are not inserted (Reported by Denis Pantsyrev)
* ASTERISK-23027 - [patch] Spelling typo "transfered" instead of
"transferred" (Reported by Jeremy Lainé)
* ASTERISK-23008 - Local channels loose CALLERID name when DAHDI
channel connects (Reported by Michael Cargile)
* ASTERISK-23100 - [patch] In chan_mgcp the ident in transmitted
request and request queue may differ - fix for locking (Reported
by adomjan)
* ASTERISK-22988 - [patch]T38 , SIP 488 after Rejecting image
media offer due to invalid or unsupported syntax (Reported by
adomjan)
* ASTERISK-22861 - [patch]Specifying a null time as parameter to
GotoIfTime or ExecIfTime causes segmentation fault (Reported by
Sebastian Murray-Roberts)
* ASTERISK-17837 - extconfig.conf - Maximum Include level (1)
exceeded (Reported by pz)
* ASTERISK-22662 - Documentation fix? - queues.conf says
persistentmembers defaults to yes, it appears to lie (Reported
by Rusty Newton)
* ASTERISK-23134 - [patch] res_rtp_asterisk port selection cannot
handle selinux port restrictions (Reported by Corey Farrell)
* ASTERISK-23220 - STACK_PEEK function with no arguments causes
crash/core dump (Reported by James Sharp)
* ASTERISK-19773 - Asterisk crash on issuing Asterisk-CLI 'reload'
command multiple times on cli_aliases (Reported by Joel Vandal)
* ASTERISK-22757 - segfault in res_clialiases.so on reload when
mapping "module reload" command (Reported by Gareth Blades)
* ASTERISK-17727 - [patch] TLS doesn't get all certificate chain
(Reported by LN)
* ASTERISK-23178 - devicestate.h: device state setting functions
are documented with the wrong return values (Reported by
Jonathan Rose)
* ASTERISK-23232 - LocalBridge AMI Event LocalOptimization value
is opposite to what's expected (Reported by Leon Roy)
* ASTERISK-23098 - [patch]possible null pointer dereference in
format.c (Reported by Marcello Ceschia)
* ASTERISK-23297 - Asterisk 12, pbx_config.so segfaults if
res_parking.so is not loaded, or if res_parking.conf has no
configuration (Reported by CJ Oster)
* ASTERISK-23069 - Custom CDR variable not recorded when set in
macro called from app_queue (Reported by Bryan Anderson)
* ASTERISK-19499 - ConfBridge MOH is not working for transferee
after attended transfer (Reported by Timo Teräs)
* ASTERISK-23261 - [patch]Output mixup in
${CHANNEL(rtpqos,audio,all)} (Reported by rsw686)
* ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic
payload change in rtp mapping in the 200 OK response (Reported
by NITESH BANSAL)
* ASTERISK-23255 - UUID included for Redhat, but missing for
Debian distros in install_prereq script (Reported by Rusty
Newton)
* ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR
variables for subsequent records (Reported by zvision)
* ASTERISK-23141 - Asterisk crashes on Dial(), in
pbx_find_extension at pbx.c (Reported by Maxim)
* ASTERISK-23336 - Asterisk warning "Don't know how to indicate
condition 33 on ooh323c" on outgoing calls from H323 to SIP peer
(Reported by Alexander Semych)
* ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
to minrate=2400, then res_fax refuse to load (Reported by David
Brillert)
* ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set
- probably introduced in 11.7.0 (Reported by OK)
* ASTERISK-23323 - [patch]chan_sip: missing p->owner checks in
handle_response_invite (Reported by Walter Doekes)
* ASTERISK-23406 - [patch]Fix typo in "sip show peer" (Reported by
ibercom)
* ASTERISK-23310 - bridged channel crashes in bridge_p2p_rtp_write
(Reported by Jeremy Lainé)
* ASTERISK-22911 - [patch]Asterisk fails to resume WebRTC call
from hold (Reported by Vytis Valentinavičius)
* ASTERISK-23104 - Specifying the SetVar AMI without a Channel
cause Asterisk to crash (Reported by Joel Vandal)
* ASTERISK-21930 - [patch]WebRTC over WSS is not working.
(Reported by John)
* ASTERISK-23383 - Wrong sense test on stat return code causes
unchanged config check to break with include files. (Reported by
David Woolley)
* ASTERISK-20149 - Crash when faxing SIP to SIP with strictrtp set
to yes (Reported by Alexandr Gordeev)
* ASTERISK-17523 - Qualify for static realtime peers does not work
(Reported by Maciej Krajewski)
* ASTERISK-21406 - [patch] chan_sip deadlock on monlock between
unload_module and do_monitor (Reported by Corey Farrell)
* ASTERISK-23373 - [patch]Security: Open FD exhaustion with
chan_sip Session-Timers (Reported by Corey Farrell)
* ASTERISK-23340 - Security Vulnerability: stack allocation of
cookie headers in loop allows for unauthenticated remote denial
of service attack (Reported by Matt Jordan)
* ASTERISK-23311 - Manager - MoH Stop Event fails to show up when
leaving Conference (Reported by Benjamin Keith Ford)
* ASTERISK-23420 - [patch]Memory leak in manager_add_filter
function in manager.c (Reported by Etienne Lessard)
* ASTERISK-23488 - Logic error in callerid checksum processing
(Reported by Russ Meyerriecks)
* ASTERISK-23461 - Only first user is muted when joining
confbridge with 'startmuted=yes' (Reported by Chico Manobela)
* ASTERISK-20841 - fromdomain not honored on outbound INVITE
request (Reported by Kelly Goedert)
* ASTERISK-22079 - Segfault: INTERNAL_OBJ (user_data=0x6374652f)
at astobj2.c:120 (Reported by Jamuel Starkey)
* ASTERISK-23509 - [patch]SayNumber for Polish language tries to
play empty files for numbers divisible by 100 (Reported by
zvision)
* ASTERISK-23103 - [patch]Crash in ast_format_cmp, in ao2_find
(Reported by JoshE)
* ASTERISK-23391 - Audit dialplan function usage of channel
variable (Reported by Corey Farrell)
* ASTERISK-23548 - POST to ARI sometimes returns no body on
success (Reported by Scott Griepentrog)
* ASTERISK-23460 - ooh323 channel stuck if call is placed directly
and gatekeeper is not available (Reported by Dmitry Melekhov)
Improvements made in this release:
-----------------------------------
* ASTERISK-22980 - [patch]Allow building cdr_radius and cel_radius
against libfreeradius-client (Reported by Jeremy Lainé)
* ASTERISK-22661 - Unable to exit ChanSpy if spied channel does
not have a call in progress (Reported by Chris Hillman)
* ASTERISK-23099 - [patch] WSS: enable ast_websocket_read()
function to read the whole available data at first and then wait
for any fragmented packets (Reported by Thava Iyer)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.9.0
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The
available security releases are released as versions 1.8.15-cert5,
11.6-cert2, 1.8.26.1, 11.8.1, and 12.1.1.
The release of these versions resolve the following issues:
* AST-2014-001: Stack overflow in HTTP processing of Cookie headers.
Sending a HTTP request that is handled by Asterisk with a large number of
Cookie headers could overflow the stack.
Another vulnerability along similar lines is any HTTP request with a
ridiculous number of headers in the request could exhaust system memory.
* AST-2014-002: chan_sip: Exit early on bad session timers request
This change allows chan_sip to avoid creation of the channel and
consumption of associated file descriptors altogether if the inbound
request is going to be rejected anyway.
These issues and their resolutions are described in the security advisories.
For more information about the details of these vulnerabilities,
please read security advisories AST-2014-001, AST-2014-002,
AST-2014-003, and AST-2014-004, which were released at the same
time as this announcement.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.8.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2014-001.pdf
* http://downloads.asterisk.org/pub/security/AST-2014-002.pdf
Thank you for your continued support of Asterisk!
-----
The Asterisk Development Team has announced the release of Asterisk 11.8.0.
The release of Asterisk 11.8.0 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-22544 - Italian prompt vm-options has advertisement in
it (Reported by Rusty Newton)
* ASTERISK-21383 - STUN Binding Requests Not Being Sent Back from
Asterisk to Chrome (Reported by Shaun Clark)
* ASTERISK-22478 - [patch]Can't use pound(hash) symbol for custom
DTMF menus in ConfBridge (processed as directive) (Reported by
Nicolas Tanski)
* ASTERISK-12117 - chan_sip creates a new local tag (from-tag) for
every register message (Reported by Pawel Pierscionek)
* ASTERISK-20862 - Asterisk min and max member penalties not
honored when set with 0 (Reported by Schmooze Com)
* ASTERISK-22746 - [patch]Crash in chan_dahdi during caller id
read (Reported by Michael Walton)
* ASTERISK-22788 - [patch] main/translate.c: access to variable f
after free in ast_translate() (Reported by Corey Farrell)
* ASTERISK-21242 - Segfault when T.38 re-invite retransmission
receives 200 OK (Reported by Ashley Winters)
* ASTERISK-22590 - BufferOverflow in unpacksms16() when receiving
16 bit multipart SMS with app_sms (Reported by Jan Juergens)
* ASTERISK-22905 - Prevent Asterisk functions that are 'dangerous'
from being executed from external interfaces (Reported by Matt
Jordan)
* ASTERISK-23021 - Typos in code : "avaliable" instead of
"available" (Reported by Jeremy Lainé)
* ASTERISK-22970 - [patch]Documentation fix for QUOTE() (Reported
by Gareth Palmer)
* ASTERISK-21960 - ooh323 channels stuck (Reported by Dmitry
Melekhov)
* ASTERISK-22350 - DUNDI - core dump on shutdown - segfault in
sqlite3_reset from /usr/lib/libsqlite3.so.0 (Reported by Birger
"WIMPy" Harzenetter)
* ASTERISK-22942 - [patch] - Asterisk crashed after
Set(FAXOPT(faxdetect)=t38) (Reported by adomjan)
* ASTERISK-22856 - [patch]SayUnixTime in polish reads minutes
instead of seconds (Reported by Robert Mordec)
* ASTERISK-22854 - [patch] - Deadlock between cel_pgsql unload and
core_event_dispatcher taskprocessor thread (Reported by Etienne
Lessard)
* ASTERISK-22910 - [patch] - REPLACE() calls strcpy on overlapping
memory when <replace-char> is empty (Reported by Gareth Palmer)
* ASTERISK-22871 - cel_pgsql module not loading after "reload" or
"reload cel_pgsql.so" command (Reported by Matteo)
* ASTERISK-23084 - [patch]rasterisk needlessly prints the
AST-2013-007 warning (Reported by Tzafrir Cohen)
* ASTERISK-17138 - [patch] Asterisk not re-registering after it
receives "Forbidden - wrong password on authentication"
(Reported by Rudi)
* ASTERISK-23011 - [patch]configure.ac and pbx_lua don't support
lua 5.2 (Reported by George Joseph)
* ASTERISK-22834 - Parking by blind transfer when lot full orphans
channels (Reported by rsw686)
* ASTERISK-23047 - Orphaned (stuck) channel occurs during a failed
SIP transfer to parking space (Reported by Tommy Thompson)
* ASTERISK-22946 - Local From tag regression with sipgate.de
(Reported by Stephan Eisvogel)
* ASTERISK-23010 - No BYE message sent when sip INVITE is received
(Reported by Ryan Tilton)
* ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set
- probably introduced in 11.7.0 (Reported by OK)
Improvements made in this release:
-----------------------------------
* ASTERISK-22728 - [patch] Improve Understanding Of 'Forcerport'
When Running "sip show peers" (Reported by Michael L. Young)
* ASTERISK-22659 - Make a new core and extra sounds release
(Reported by Rusty Newton)
* ASTERISK-22919 - core show channeltypes slicing (Reported by
outtolunc)
* ASTERISK-22918 - dahdi show channels slices PRI channel dnid on
output (Reported by outtolunc)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.8.0
Thank you for your continued support of Asterisk!
Changes since version 0.4:
0.7, 20130330 - jeagle
Add ability to allow users to specify their own frame allocation routines.
Update API mode 2 with latest version from jdodgen
0.6, 20120624 - jeagle
Update documentation.
Add support for API mode 2 escapes. Needs testing.
Add constant for the "BD" baud rate table.
0.5, 20120401 - jeagle
Add support for Win32::SerialPort to enable Windows support. (Thanks Jerry)
Fix issue with tx() in async mode. (Thanks Vicente)
Add support for "explicit rx indicator" packets. (Thanks Vicente)
Do it for all packages that
* mention perl, or
* have a directory name starting with p5-*, or
* depend on a package starting with p5-
like last time, for 5.18, where this didn't lead to complaints.
Let me know if you have any this time.
These releases are maintenance releases, and do not contain any new
features or functionality, but only contain bugfixes:
* Re-order library files in fchmod() configure check
* faxalter: Wire up the nissing page range -Z options
* man: JobReqError/JobRetryError were missing in hylafax-config.4
* typerules: adding missing comma to typeNames array
* Do not warn about one of the Fontpath directories not existing
* Reworked how faxsetup looks for Fontmap
* Use a private Fontmap.HylaFAX file of .pfb files
* Combine all Fontmap files in memory, including new Fontmap.HylaFAX
* Bug 934: We need to avoid a 0-index in playList
* hfaxd: Eliminte extraneous debug logging
* hfaxd: Make source port for active connections be ctrl port - 1
* hfaxd: Release old accept fd
* Support libtiff 4.0
* faxsend: JobRetryOther/JobRequeueOther weren't actually being used
* Make sure not to cut faxq FIFO messages in two when reaching end of buffer
* hfaxd: Port is network byte order, correct logging of it
This fixes abuild failure when a version of the package is already
installed.
Not bumping PKGREVISION because the resulting package should be unchanged.
Revision history for Perl extension Device::Modem.
1.57 Sun Jan 26 11:36:11 CET 2014
- Added a "handshake" option to the connect() method.
Allowed values are "xoff", "rts" or "none" (default).
Thanks to Ezio Bonsi for suggesting the idea.
either because they themselves are not ready or because a
dependency isn't. This is annotated by
PYTHON_VERSIONS_INCOMPATIBLE= 33 # not yet ported as of x.y.z
or
PYTHON_VERSIONS_INCOMPATIBLE= 33 # py-foo, py-bar
respectively, please use the same style for other packages,
and check during updates.
Use versioned_dependencies.mk where applicable.
Use REPLACE_PYTHON instead of handcoded alternatives, where applicable.
Reorder Makefile sections into standard order, where applicable.
Remove PYTHON_VERSIONS_INCLUDE_3X lines since that will be default
with the next commit.
Whitespace cleanups and other nits corrected, where necessary.
Version 2.7 2013-10-17
---------------------------
- Win32: setRTS and setDTR can be called before the port is opened and it will
set the initial state on port open.
- Posix: add platform specific method: outWaiting (already present for Win32)
- Posix: rename flowControl to setXON to match name on Win32, add
flowControlOut function
- rfc2217: zero polls value (baudrate, data size, stop bits, parity) (Erik
Lundh)
- Posix: [Patch pyserial:28] Accept any speed on Linux [update]
- Posix: [Patch pyserial:29] PosixSerial.read() should "ignore" errno.EINTR
- OSX: [Patch pyserial:27] Scan by VendorID/Product ID for USB Serial devices
- Ensure working with bytes in write() calls
Bugfixes:
- [Bug 3540332] SerialException not returned
- [Bug pyserial:145] Error in socket_connection.py
- [Bug pyserial:135] reading from socket with timeout=None causes TypeError
- [Bug pyserial:130] setup.py should not append py3k to package name
- [Bug pyserial:117] no error on lost conn w/socket://
Bugfixes (posix):
- [Patch 3462364] Fix: NameError: global name 'base' is not defined
- list_ports and device() for BSD updated (Anders Langworthy)
- [Bug 3518380] python3.2 -m serial.tools.list_ports error
- [Bug pyserial:137] Patch to add non-standard baudrates to Cygwin
- [Bug pyserial:141] open: Pass errno from IOError to SerialException
- [Bug pyserial:125] Undefined 'base' on list_ports_posix.py, function usb_lsusb
- [Bug pyserial:151] Serial.write() without a timeout uses 100% CPU on POSIX
- [Patch pyserial:30] [PATCH 1/1] serial.Serial() should not raise IOError.
Bugfixes (win32):
- [Bug 3444941] ctypes.WinError() unicode error
- [Bug 3550043] on Windows in tools global name 'GetLastError' is not defined
- [Bug pyserial:146] flush() does nothing in windows (despite docs)
- [Bug pyserial:144] com0com ports ignored due to missing "friendly name"
- [Bug pyserial:152] Cannot configure port, some setting was wrong. Can leave
port handle open but port not accessible
The Asterisk Development Team has announced the release of Asterisk 1.8.25.0.
The release of Asterisk 1.8.25.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* --- chan_sip: Fix an issue where an incompatible audio format may be
added to SDP.
* --- cdr_adaptive_odbc: Also apply a filter when the CDR value is
empty.
* --- app_queue: Fix Queuelog EXITWITHKEY only logging two of four
fields
* --- chan_sip: Do not increment the SDP version between 183 and 200
responses.
* --- chan_sip: Fix Realtime Peer Update Problem When Un-registering
And Expires Header In 200ok
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.25.0
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 11.7.0.
The release of Asterisk 11.7.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* --- app_confbridge: Can now set the language used for announcements
to the conference.
* --- app_queue: Fix CLI "queue remove member" queue_log entry.
* --- chan_sip: Do not increment the SDP version between 183 and 200
responses.
* --- chan_sip: Allow a sip peer to accept both AVP and AVPF calls
* --- chan_sip: Fix Realtime Peer Update Problem When Un-registering
And Expires Header In 200ok
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.7.0
Thank you for your continued support of Asterisk!
