The Asterisk Development Team has announced the release of Asterisk 11.3.0.
The release of Asterisk 11.3.0 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:
* --- Fix issue where chan_mobile fails to bind to first available port
* --- Fix Queue Log Reporting Every Call COMPLETECALLER With "h"
Extension Present
* --- Retain XMPP filters across reconnections so external modules
continue to function as expected.
* --- Ensure that a declined media stream is terminated with a '\r\n'
* --- Fix pjproject compilation in certain circumstances
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.3.0
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 1.8.21.0.
The release of Asterisk 1.8.21.0 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:
* --- Fix issue where chan_mobile fails to bind to first available port
* --- Fix station ringback; trunk hangup issues in SLA
* --- Fix Queue Log Reporting Every Call COMPLETECALLER With "h"
Extension Present
* --- Fix Record-Route parsing for large headers.
* --- Fix AMI redirect action with two channels failing to redirect
both channels.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.21.0
Thank you for your continued support of Asterisk!
AST-2013-001, AST-2013-002, and AST-2013-003.
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.15 and Asterisk 1.8, 10, and 11. The available security releases
are released as versions 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones,
and 11.2.2.
The release of these versions resolve the following issues:
* A possible buffer overflow during H.264 format negotiation. The format
attribute resource for H.264 video performs an unsafe read against a media
attribute when parsing the SDP.
This vulnerability only affected Asterisk 11.
* A denial of service exists in Asterisk's HTTP server. AST-2012-014, fixed
in January of this year, contained a fix for Asterisk's HTTP server for a
remotely-triggered crash. While the fix prevented the crash from being
triggered, a denial of service vector still exists with that solution if an
attacker sends one or more HTTP POST requests with very large Content-Length
values.
This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11
* A potential username disclosure exists in the SIP channel driver. When
authenticating a SIP request with alwaysauthreject enabled, allowguest
disabled, and autocreatepeer disabled, Asterisk discloses whether a user
exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways.
This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11
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-2013-001, AST-2013-002, and AST-2013-003, 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-11.2.2
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2013-001.pdf
* http://downloads.asterisk.org/pub/security/AST-2013-002.pdf
* http://downloads.asterisk.org/pub/security/AST-2013-003.pdf
Thank you for your continued support of Asterisk!
AST-2013-001, AST-2013-002, and AST-2013-003.
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.15 and Asterisk 1.8, 10, and 11. The available security releases
are released as versions 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones,
and 11.2.2.
The release of these versions resolve the following issues:
* A denial of service exists in Asterisk's HTTP server. AST-2012-014, fixed
in January of this year, contained a fix for Asterisk's HTTP server for a
remotely-triggered crash. While the fix prevented the crash from being
triggered, a denial of service vector still exists with that solution if an
attacker sends one or more HTTP POST requests with very large Content-Length
values.
This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11
* A potential username disclosure exists in the SIP channel driver. When
authenticating a SIP request with alwaysauthreject enabled, allowguest
disabled, and autocreatepeer disabled, Asterisk discloses whether a user
exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways.
This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11
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-2013-001, AST-2013-002, and AST-2013-003, 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-10.12.2
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2013-001.pdf
* http://downloads.asterisk.org/pub/security/AST-2013-002.pdf
* http://downloads.asterisk.org/pub/security/AST-2013-003.pdf
Thank you for your continued support of Asterisk!
AST-2013-001, AST-2013-002, and AST-2013-003.
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.15 and Asterisk 1.8, 10, and 11. The available security releases
are released as versions 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones,
and 11.2.2.
The release of these versions resolve the following issues:
* A denial of service exists in Asterisk's HTTP server. AST-2012-014, fixed
in January of this year, contained a fix for Asterisk's HTTP server for a
remotely-triggered crash. While the fix prevented the crash from being
triggered, a denial of service vector still exists with that solution if an
attacker sends one or more HTTP POST requests with very large Content-Length
values.
This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11
* A potential username disclosure exists in the SIP channel driver. When
authenticating a SIP request with alwaysauthreject enabled, allowguest
disabled, and autocreatepeer disabled, Asterisk discloses whether a user
exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways.
This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11
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-2013-001, AST-2013-002, and AST-2013-003, 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-1.8.20.2
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2013-001.pdf
* http://downloads.asterisk.org/pub/security/AST-2013-002.pdf
* http://downloads.asterisk.org/pub/security/AST-2013-003.pdf
Thank you for your continued support of Asterisk!
File too long (should be no more than 24 lines).
Line too long (should be no more than 80 characters).
Trailing empty lines.
Trailing white-space.
Trucated the long files as best as possible while preserving the most info
contained in them.
pkglint warnings aren't gospel! They need to be verified in an
intelligent manner. After variable substitution, the lines will
be shorter then 80 characters, thus there was no need to shorten
them.
COMMENT should not be longer than 70 characters.
COMMENT should not begin with 'A'.
COMMENT should not begin with 'An'.
COMMENT should not begin with 'a'.
COMMENT should not end with a period.
COMMENT should start with a capital letter.
pkglint warnings. Some files also got minor formatting, spelling, and style
corrections.
There is a new maintainer, Hendrik Sattler and the sources are held
at gitorius.org. Build is changed to CMake and although it does not appear
in the ChangeLog below, I forwarded all the NetBSD related patches that
pkgsrc had (and more in fact, to get it working) although I would still say
that obexapp is the better program.
From the ChangeLog:
ObexFTP 0.24 (released 2013-03-05)
----------------------------------
* Maintainer changed from Christian to Hendrik
* change build system to using CMake
* integrate obexfs-0.12
* fix build for OpenOBEX-1.7
ObexFTP 0.23 (released 2009-02-17)
----------------------------------
* allow win32 to use hci src names
* adding a simpler connect wrapper
* show OBEX_HandleInput errors
* catch errors and let the user know
* print timeout stats if available
* sdp unregister more verbose
* upgrading btkit
* fix for win32 without bt
* replacing deprecated automake vars
* sizeof() fixes
* removing bdaddr_t reference from obexftpd.c
* adding bootstrap helper
* concurrency bug in extconf.rb generated Makefile (fix by Alin Năstac)
* switching from POD to asciidoc
* clearing gnu-style implicit rules
ObexFTP 0.22 (released 2008-06-15)
----------------------------------
* added proper unicode support
* added support for transparent OBEX-over-AT mode
* rewritten at-command function
* added specific error messages
* refactored to flexible bt_kit layer
* fixed cache root duplicates
* fixed off-by-one and unfreed mem in cache layer
* added pkg-config file
* added example code
* switched to doxygen
* added python binding callbacks
* portable packed structs
* enabled linux hci dev names for source selection
* Python binding uses distutils now, tested by Adam Williamson
* removed exit from bt discovery
* Better autodetection for possible language bindings
* reworked win32 support
* Motorola SLVR L2 cobex fix by Andrey Rahmatullin
* now using AC_HELP_STRING for compat with autoconf <=2.57
* added hci selection support, drafted by Manuel Naranjo
* switched obexftp cli to new discovery api
* prefer PCSUITE over FTP, req. by Martin Storsjö for Series 60 2nd Ed.
* fixed compile error with >=swig-1.3.28
* renamed sdp browse function
* fixed month/day swapping in atotime, spotted by Dr. Johannes Zellner
* added BFC compatibility for newer Siemens phones
* added PCSOFTWARE uuid support for SHARP phones
* added motorola support
* end bfb mode properly
* added CPROT=0 support from 3GPP 27.007
* fixed ericsson init
* fixed invalid conn_id in disconnect rep. by Alan J. McFarlane
* better create flag handling in setpath
* Changed LDADD to LIBADD sug. by Sergey Vlasov <vsu@altlinux.ru>
* obexftpd clean up by Hendrik Sattler
* 64-bit fixes by Hendrik Sattler
* Removed all (dangerous) obex_headerdata_t casts
* Reorganized all swig-dependant Makefiles
* Applied cobex write patch from Simon Ruggier <Simon80@gmail.com>
* Applied from Frode Isaksen <fisaksen@bewan.com>
ObexFTP 0.21 (released 2006-06-27)
----------------------------------
2006-05-26 Christian W. Zuckschwerdt <zany@triq.net>
* Fixes to obexftpd suggested by Hendrik Sattler
2006-05-24 Christian W. Zuckschwerdt <zany@triq.net>
* Added ruby binding
* Added preliminary discovery function
and the sources are now stored at gitorius.org. The build system is changed
to CMake
From the ChangeLog:
ver 1.7:
Add support for CMake config files
Internal code reorganisation and rewrite
Add new function set for better control than OBEX_HandleInput():
* OBEX_SetTimeout(),
* OBEX_Work() and
* OBEX_GetDataDirection()
ver 1.6:
Change ABI from 1 to 2 because:
* Redo the USB changes from version 1.4
* Remove InOBEX_* function, use the TcpOBEX_* functions instead
* Remove the simple Unicode<->ASCII functions
Add support for Single Response Mode
Add manpages for all example applications
Add udev support
Add new example app to find IrDA and USB OBEX devices
Add fixes for FreeBSD
Add support for libusb-1.x
Add support for close-on-exec
RXTX is a Java library, using a native implementation (via JNI), providing
serial and parallel communication for the Java Development Toolkit (JDK).
It is based on the specification for Sun's Java Communications API, though
while many of the class descriptions are the same the package used it not,
since gnu.io is used instead. A certain amount of compatibility is intended
with API, though this project should be considered as a fork and therefore
compatible in spirit, but not in implementation.
----- 11.2.1:
The Asterisk Development Team has announced the release of Asterisk 11.2.1.
The release of Asterisk 11.2.1 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:
* --- Fix astcanary startup problem due to wrong pid value from before
daemon call
* --- Update init.d scripts to handle stderr; readd splash screen for
remote consoles
* --- Reset RTP timestamp; sequence number on SSRC change
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.2.1
Thank you for your continued support of Asterisk!
----- 11.2.0:
The Asterisk Development Team has announced the release of Asterisk 11.2.0.
The release of Asterisk 11.2.0 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_meetme: Fix channels lingering when hung up under certain
conditions
* --- Fix stuck DTMF when bridge is broken.
* --- Add missing support for "who hung up" to chan_motif.
* --- Remove a fixed size limitation for producing SDP and change how
ICE support is disabled by default.
* --- Fix chan_sip websocket payload handling
* --- Fix pjproject compilation in certain circumstances
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.2.0
Thank you for your continued support of Asterisk!
----- 10.12.1
The Asterisk Development Team has announced the release of Asterisk 10.12.1.
The release of Asterisk 10.12.1 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:
* --- Fix astcanary startup problem due to wrong pid value from before
daemon call
* --- Update init.d scripts to handle stderr; readd splash screen for
remote consoles
* --- Reset RTP timestamp; sequence number on SSRC change
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.12.1
Thank you for your continued support of Asterisk!
----- 10.12.0
The Asterisk Development Team has announced the release of Asterisk 10.12.0.
The release of Asterisk 10.12.0 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_meetme: Fix channels lingering when hung up under certain
conditions
* --- Fix stuck DTMF when bridge is broken.
* --- Improve Code Readability And Fix Setting natdetected Flag
* --- Fix extension matching with the '-' char.
* --- Fix call files when astspooldir is relative.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.12.0
Thank you for your continued support of Asterisk!
----- 1.8.20.1
The Asterisk Development Team has announced the release of Asterisk 1.8.20.1.
The release of Asterisk 1.8.20.1 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:
* --- Fix astcanary startup problem due to wrong pid value from before
daemon call
* --- Update init.d scripts to handle stderr; readd splash screen for
remote consoles
* --- Reset RTP timestamp; sequence number on SSRC change
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.20.1
Thank you for your continued support of Asterisk!
----- 1.8.20.0
The Asterisk Development Team has announced the release of Asterisk 1.8.20.0.
The release of Asterisk 1.8.20.0 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_meetme: Fix channels lingering when hung up under certain
conditions
* --- Fix stuck DTMF when bridge is broken.
* --- Improve Code Readability And Fix Setting natdetected Flag
* --- Fix extension matching with the '-' char.
* --- Fix call files when astspooldir is relative.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.20.0
Thank you for your continued support of Asterisk!
and AST-2012-015. Apparently the last update didn't completely
fix the issues.
The Asterisk Development Team has announced a security release for
Asterisk 11, Asterisk 11.1.2. This release addresses the security
vulnerabilities reported in AST-2012-014 and AST-2012-015, and
replaces the previous version of Asterisk 11 released for these
security vulnerabilities. The prior release left open a vulnerability
in res_xmpp that exists only in Asterisk 11; as such, other versions
of Asterisk were resolved correctly by the previous releases.
The release of these versions resolve the following two issues:
* Stack overflows that occur in some portions of Asterisk that manage a TCP
connection. In SIP, this is exploitable via a remote unauthenticated session;
in XMPP and HTTP connections, this is exploitable via remote authenticated
sessions. The vulnerabilities in SIP and HTTP were corrected in a prior
release of Asterisk; the vulnerability in XMPP is resolved in this release.
* A denial of service vulnerability through exploitation of the device state
cache. Anonymous calls had the capability to create devices in Asterisk that
would never be disposed of. Handling the cachability of device states
aggregated via XMPP is handled in this release.
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-2012-014 and AST-2012-015.
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.1.2
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2012-014.pdf
* http://downloads.asterisk.org/pub/security/AST-2012-015.pdf
Thank you for your continued support of Asterisk - and we apologize for having
to do this twice!
and AST-2012-015.
Approved for commit during freeze by: agc
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.11 and Asterisk 1.8, 10, and 11. The available security releases
are released as versions 1.8.11-cert10, 1.8.19.1, 10.11.1, 10.11.1-digiumphones,
and 11.1.1.
The release of these versions resolve the following two issues:
* Stack overflows that occur in some portions of Asterisk that manage a TCP
connection. In SIP, this is exploitable via a remote unauthenticated session;
in XMPP and HTTP connections, this is exploitable via remote authenticated
sessions.
* A denial of service vulnerability through exploitation of the device state
cache. Anonymous calls had the capability to create devices in Asterisk that
would never be disposed of.
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-2012-014 and AST-2012-015, 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-11.1.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2012-014.pdf
* http://downloads.asterisk.org/pub/security/AST-2012-015.pdf
Thank you for your continued support of Asterisk!
and AST-2012-015.
Approved for commit during freeze by: agc
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.11 and Asterisk 1.8, 10, and 11. The available security releases
are released as versions 1.8.11-cert10, 1.8.19.1, 10.11.1, 10.11.1-digiumphones,
and 11.1.1.
The release of these versions resolve the following two issues:
* Stack overflows that occur in some portions of Asterisk that manage a TCP
connection. In SIP, this is exploitable via a remote unauthenticated session;
in XMPP and HTTP connections, this is exploitable via remote authenticated
sessions.
* A denial of service vulnerability through exploitation of the device state
cache. Anonymous calls had the capability to create devices in Asterisk that
would never be disposed of.
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-2012-014 and AST-2012-015, 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-11.1.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2012-014.pdf
* http://downloads.asterisk.org/pub/security/AST-2012-015.pdf
Thank you for your continued support of Asterisk!
and AST-2012-015.
Approved for commit during freeze by: agc
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.11 and Asterisk 1.8, 10, and 11. The available security releases
are released as versions 1.8.11-cert10, 1.8.19.1, 10.11.1, 10.11.1-digiumphones,
and 11.1.1.
The release of these versions resolve the following two issues:
* Stack overflows that occur in some portions of Asterisk that manage a TCP
connection. In SIP, this is exploitable via a remote unauthenticated session;
in XMPP and HTTP connections, this is exploitable via remote authenticated
sessions.
* A denial of service vulnerability through exploitation of the device state
cache. Anonymous calls had the capability to create devices in Asterisk that
would never be disposed of.
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-2012-014 and AST-2012-015, 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-11.1.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2012-014.pdf
* http://downloads.asterisk.org/pub/security/AST-2012-015.pdf
Thank you for your continued support of Asterisk!
Note that Asterisk 10.* will be going into security fix only mode
on Dec. 15th, 2012. Users may wish to consider moving to one of
the Long Term Support versions: comms/asterisk18 (Asterisk 1.8.*)
or comms/asterisk (which currently has Asterisk 11.*). See
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions for
information on Asterisk versions.
----- 10.11.0:
The Asterisk Development Team has announced the release of Asterisk 10.11.0.
The release of Asterisk 10.11.0 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:
* --- Prevent resetting of NATted realtime peer address on reload.
* --- Do not use a FILE handle when doing SIP TCP reads.
* --- Fix ConfBridge crash if no timing module loaded.
* --- confbridge: Fix a bug which made conferences not record with
AMI/CLI commands
* --- Fix execution of 'i' extension due to uninitialized variable.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.11.0
Thank you for your continued support of Asterisk!
----- 10.10.1:
The Asterisk Development Team has announced the release of Asterisk 10.10.1.
The release of Asterisk 10.10.1 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:
* --- chan_local: Fix local_pvt ref leak in local_devicestate().
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.10.1
Thank you for your continued support of Asterisk!
----- 1.8.19.0:
The Asterisk Development Team has announced the release of Asterisk 1.8.19.0.
The release of Asterisk 1.8.19.0 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:
* --- Prevent resetting of NATted realtime peer address on reload.
* --- Do not use a FILE handle when doing SIP TCP reads.
* --- Fix execution of 'i' extension due to uninitialized variable.
* --- Ensure that the Queue application tracks busy members in off
nominal situations
* --- Properly extract the Body information of an EWS calendar item
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.19.0
Thank you for your continued support of Asterisk!
----- 1.8.18.1:
The Asterisk Development Team has announced the release of Asterisk 1.8.18.1.
The release of Asterisk 1.8.18.1 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:
* --- chan_local: Fix local_pvt ref leak in local_devicestate().
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.18.1
Thank you for your continued support of Asterisk!
Relevant ChangeLog entries since 2.5:
* src/main.c, src/minicom.c: iconv: Handle the case that iconv
did not convert anything. Reported by Mike Crowe, Debian #659351.
* src/ipc.c: Formatting cleanup.
* src/main.c: ETIME -> ETIMEDOUT as ETIME is not available on BSDs
* src/main.c: Fix invalid memory used, reported by Larry Baker
* src/config.c, src/rwconf.c: Do not set modem init and reset string
anymore, define them empty. Instead, when editing those offer
them as a default.
* src/minicom.h, src/main.c, src/dial.c: only update statusline
if there's a change (e.g. for updates times)
* src/updown.c: Flush before forking helper program,
patch by Domen Puncer, thanks!
* src/minicom.c, src/minicom.h, src/vt100.c: Add timestamps with
milliseconds, based on patch by Rapha�l Ass�nat, thanks!
* src/dial.c, src/minicom.c, src/main.c: Cleanups. Print
basename of current device to statusline if online time is disabled.
* configure.in, src/Makefile.am, src/main.c, src/minicom.c,
src/minicom.h, src/updown.c: Add lockdev support,
by Ludwig Nussel <ludwig.nussel@suse.de>
* src/dial.c: add a dialdir version 6 which does not save the
pointer on disk and should now work on 32 and 64 bit
systems equally.
* configure.in: Use AM_ICONV_LINK...
* src/script.c: Fix a buffer overflow problem. Thanks Frederic Germain.
* src/minicom.c: Do not use iconv-functions if iconv is not available.
* src/config.c, src/main.c, src/vt100.c, src/vt100.h: Add transmit
delay for every character, based on patch by Nicolas PILLON.
* src/config.c: Do not extend tilde to home directory for
non-path arguments. Debian bug #621741
* configure.in, src/Makefile.am: Add workaround and then use
libiconv for linking, fixes build issue on Mac OS X.
* src/main.c: Increase serial port open timeout, by
Lubomir Rintel
* src/main.c: Set sensible errno if port open times out,
by Lubomir Rintel
* src/help.c: Help fix for timestamp toggle by Mark Einon
* src/minicom.c: Code consolidation.
* src/minicom.c, src/minicom.h, src/vt100.c, man/minicom.1: Make
line timestamps three value: every line, every second, and off.
* man/minicom.1: Wording fix.
* src/vt100.c: Cleanups: Delete everything that was in OLD blocks.
Do not explicitly set global variables to 0.
* src/vt100.c, man/minicom.1: Change timestamp style, prepend every
line. Add in manpage.
* src/dial.c src/help.c src/ipc.c src/minicom.c src/minicom.h
src/vt100.c src/vt100.h: Addition by Mark Einon
<mark.einon@gmail.com> to add current date/time to each line.
* src/file.c: Only enter directory if we have read permissions to
get directory listings, by Jan Görig.
* src/file.c, src/getsdir.c: Cleanup and simplify.
* man/runscript.1, man/minicom.1: Fixes by John Bradshaw
* src/main.c: Avoid redraw of status line in Offline mode when
nothing changed.
* src/minicom.c: Do not lose line wrap setting over terminal resizes.
* src/main.c: Simplify status line update, also makes status
messages display the amount of time they are actually supposed
to display.
