diff --git a/.version b/.version index ad0fe45646..e3154eb566 100644 --- a/.version +++ b/.version @@ -1 +1 @@ -18.26.1 +18.26.2 diff --git a/CHANGES.html b/CHANGES.html new file mode 120000 index 0000000000..23539d495e --- /dev/null +++ b/CHANGES.html @@ -0,0 +1 @@ +ChangeLogs/ChangeLog-18.26.2.html \ No newline at end of file diff --git a/CHANGES.md b/CHANGES.md index 13bd117716..35aabbf103 120000 --- a/CHANGES.md +++ b/CHANGES.md @@ -1 +1 @@ -ChangeLogs/ChangeLog-18.26.1.md \ No newline at end of file +ChangeLogs/ChangeLog-18.26.2.md \ No newline at end of file diff --git a/ChangeLogs/ChangeLog-18.26.2.html b/ChangeLogs/ChangeLog-18.26.2.html new file mode 100644 index 0000000000..449acff84d --- /dev/null +++ b/ChangeLogs/ChangeLog-18.26.2.html @@ -0,0 +1,66 @@ +
Author: George Joseph + Date: 2025-05-19
+UserNote: A new asterisk.conf option 'disable_remote_console_shell' has + been added that, when set, will prevent remote consoles from executing + shell commands using the '!' prefix.
+Resolves: #GHSA-c7p6-7mvq-8jq2
+Author: George Joseph + Date: 2025-03-24
+Incoming SIP MESSAGEs will now have their From header's display name + sanitized by replacing any characters < 32 (space) with a space.
+Resolves: #GHSA-2grh-7mhv-fcfw
+ diff --git a/ChangeLogs/ChangeLog-18.26.2.md b/ChangeLogs/ChangeLog-18.26.2.md new file mode 100644 index 0000000000..b9bad9346c --- /dev/null +++ b/ChangeLogs/ChangeLog-18.26.2.md @@ -0,0 +1,75 @@ + +## Change Log for Release asterisk-18.26.2 + +### Links: + + - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html) + - [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2) + - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz) + - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) + +### Summary: + +- Commits: 2 +- Commit Authors: 1 +- Issues Resolved: 0 +- Security Advisories Resolved: 2 + - [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw): Using malformed From header can forge identity with ";" or NULL in name portion + - [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2): cli_permissions.conf: deny option does not work for disallowing shell commands + +### User Notes: + +- #### asterisk.c: Add option to restrict shell access from remote consoles. + A new asterisk.conf option 'disable_remote_console_shell' has + been added that, when set, will prevent remote consoles from executing + shell commands using the '!' prefix. + Resolves: #GHSA-c7p6-7mvq-8jq2 + + +### Upgrade Notes: + + +### Commit Authors: + +- George Joseph: (2) + +## Issue and Commit Detail: + +### Closed Issues: + + - !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";" or NULL in name portion + - !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for disallowing shell commands + +### Commits By Author: + +- #### George Joseph (2): + - res_pjsip_messaging.c: Mask control characters in received From display name + - asterisk.c: Add option to restrict shell access from remote consoles. + + +### Commit List: + +- asterisk.c: Add option to restrict shell access from remote consoles. +- res_pjsip_messaging.c: Mask control characters in received From display name + +### Commit Details: + +#### asterisk.c: Add option to restrict shell access from remote consoles. + Author: George Joseph + Date: 2025-05-19 + + UserNote: A new asterisk.conf option 'disable_remote_console_shell' has + been added that, when set, will prevent remote consoles from executing + shell commands using the '!' prefix. + + Resolves: #GHSA-c7p6-7mvq-8jq2 + +#### res_pjsip_messaging.c: Mask control characters in received From display name + Author: George Joseph + Date: 2025-03-24 + + Incoming SIP MESSAGEs will now have their From header's display name + sanitized by replacing any characters < 32 (space) with a space. + + Resolves: #GHSA-2grh-7mhv-fcfw + diff --git a/README.html b/README.html new file mode 100644 index 0000000000..482f264bf3 --- /dev/null +++ b/README.html @@ -0,0 +1,205 @@ + By Mark Spencer <markster@digium.com> and the Asterisk.org developer community.
+ Copyright (C) 2001-2021 Sangoma Technologies Corporation and other copyright holders.
+
+It is imperative that you read and fully understand the contents of +the security information document before you attempt to configure and run +an Asterisk server.
+See Important Security Considerations for more information.
+Asterisk is an Open Source PBX and telephony toolkit. It is, in a +sense, middleware between Internet and telephony channels on the bottom, +and Internet and telephony applications at the top. However, Asterisk supports +more telephony interfaces than just Internet telephony. Asterisk also has a +vast amount of support for traditional PSTN telephony, as well.
+For more information on the project itself, please visit the Asterisk +home page and the official documentation. In addition you'll find lots +of information compiled by the Asterisk community at voip-info.org.
+There is a book on Asterisk published by O'Reilly under the Creative Commons +License. It is available in book stores as well as in a downloadable version on +the asteriskdocs.org web site.
+The Asterisk Open Source PBX is developed and tested primarily on the +GNU/Linux operating system, and is supported on every major GNU/Linux +distribution.
+Asterisk has also been 'ported' and reportedly runs properly on other +operating systems as well, including Sun Solaris, Apple's Mac OS X, Cygwin, +and the BSD variants.
+First, be sure you've got supported hardware (but note that you don't need +ANY special hardware, not even a sound card) to install and run Asterisk.
+Supported telephony hardware includes: +* All Analog and Digital Interface cards from Sangoma +* QuickNet Internet PhoneJack and LineJack +* any full duplex sound card supported by ALSA, OSS, or PortAudio +* any ISDN card supported by mISDN on Linux +* The Xorcom Astribank channel bank +* VoiceTronix OpenLine products
+If you are updating from a previous version of Asterisk, make sure you +read the UPGRADE.txt file in the source directory. There are some files +and configuration options that you will have to change, even though we +made every effort possible to maintain backwards compatibility.
+In order to discover new features to use, please check the configuration +examples in the configs directory of the source code distribution. For a +list of new features in this version of Asterisk, see the CHANGES file.
+Ensure that your system contains a compatible compiler and development +libraries. Asterisk requires either the GNU Compiler Collection (GCC) version +4.1 or higher, or a compiler that supports the C99 specification and some of +the gcc language extensions. In addition, your system needs to have the C +library headers available, and the headers and libraries for ncurses.
+There are many modules that have additional dependencies. To see what
+libraries are being looked for, see ./configure --help
, or run
+make menuselect
to view the dependencies for specific modules.
On many distributions, these dependencies are installed by packages with names +like 'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' +or similar.
+So, let's proceed: +1. Read this file.
+There are more documents than this one in the doc directory. You may also +want to check the configuration files that contain examples and reference +guides in the configs directory.
+./configure
Execute the configure script to guess values for system-dependent
+variables used during compilation. If the script indicates that some required
+components are missing, you can run ./contrib/scripts/install_prereq install
+to install the necessary components. Note that this will install all dependencies for every functionality of Asterisk. After running the script, you will need
+to rerun ./configure
.
make menuselect
[optional]This is needed if you want to select the modules that will be compiled and to +check dependencies for various optional modules.
+make
Assuming the build completes successfully:
+make install
If this is your first time working with Asterisk, you may wish to install +the sample PBX, with demonstration extensions, etc. If so, run:
+make samples
Doing so will overwrite any existing configuration files you have installed.
+ # asterisk -vvvc
+
+You'll see a bunch of verbose messages fly by your screen as Asterisk +initializes (that's the "very very verbose" mode). When it's ready, if +you specified the "c" then you'll get a command line console, that looks +like this:
+ *CLI>
+
+You can type "core show help" at any time to get help with the system. For help
+with a specific command, type "core show help
"man asterisk" at the Unix/Linux command prompt will give you detailed +information on how to start and stop Asterisk, as well as all the command +line options for starting Asterisk.
+Feel free to look over the configuration files in /etc/asterisk
, where you
+will find a lot of information about what you can do with Asterisk.
All Asterisk configuration files share a common format. Comments are +delimited by ';' (since '#' of course, being a DTMF digit, may occur in +many places). A configuration file is divided into sections whose names +appear in []'s. Each section typically contains two types of statements, +those of the form 'variable = value', and those of the form 'object => +parameters'. Internally the use of '=' and '=>' is exactly the same, so +they're used only to help make the configuration file easier to +understand, and do not affect how it is actually parsed.
+Entries of the form 'variable=value' set the value of some parameter in +asterisk. For example, in chan_dahdi.conf, one might specify:
+ switchtype=national
+
+In order to indicate to Asterisk that the switch they are connecting to is +of the type "national". In general, the parameter will apply to +instantiations which occur below its specification. For example, if the +configuration file read:
+ switchtype = national
+ channel => 1-4
+ channel => 10-12
+ switchtype = dms100
+ channel => 25-47
+
+The "national" switchtype would be applied to channels one through +four and channels 10 through 12, whereas the "dms100" switchtype would +apply to channels 25 through 47.
+The "object => parameters" instantiates an object with the given +parameters. For example, the line "channel => 25-47" creates objects for +the channels 25 through 47 of the card, obtaining the settings +from the variables specified above.
+Those using SIP phones should be aware that Asterisk is sensitive to +large jumps in time. Manually changing the system time using date(1) +(or other similar commands) may cause SIP registrations and other +internal processes to fail. If your system cannot keep accurate time +by itself use NTP to keep the system clock +synchronized to "real time". NTP is designed to keep the system clock +synchronized by speeding up or slowing down the system clock until it +is synchronized to "real time" rather than by jumping the time and +causing discontinuities. Most Linux distributions include precompiled +versions of NTP. Beware of some time synchronization methods that get +the correct real time periodically and then manually set the system +clock.
+Apparent time changes due to daylight savings time are just that, +apparent. The use of daylight savings time in a Linux system is +purely a user interface issue and does not affect the operation of the +Linux kernel or Asterisk. The system clock on Linux kernels operates +on UTC. UTC does not use daylight savings time.
+Also note that this issue is separate from the clocking of TDM +channels, and is known to at least affect SIP registrations.
+Depending on the size of your system and your configuration, +Asterisk can consume a large number of file descriptors. In UNIX, +file descriptors are used for more than just files on disk. File +descriptors are also used for handling network communication +(e.g. SIP, IAX2, or H.323 calls) and hardware access (e.g. analog and +digital trunk hardware). Asterisk accesses many on-disk files for +everything from configuration information to voicemail storage.
+Most systems limit the number of file descriptors that Asterisk can +have open at one time. This can limit the number of simultaneous +calls that your system can handle. For example, if the limit is set +at 1024 (a common default value) Asterisk can handle approximately 150 +SIP calls simultaneously. To change the number of file descriptors +follow the instructions for your system below:
+If your system uses PAM (Pluggable Authentication Modules) edit
+/etc/security/limits.conf
. Add these lines to the bottom of the file:
root soft nofile 4096
+root hard nofile 8196
+asterisk soft nofile 4096
+asterisk hard nofile 8196
+
+(adjust the numbers to taste). You may need to reboot the system for +these changes to take effect.
+If there are no instructions specifically adapted to your system
+above you can try adding the command ulimit -n 8192
to the script
+that starts Asterisk.
See the doc directory for more documentation on various features. +Again, please read all the configuration samples that include documentation +on the configuration options.
+Finally, you may wish to visit the support site and join the mailing +list if you're interested in getting more information.
+Welcome to the growing worldwide community of Asterisk users!
+ Mark Spencer, and the Asterisk.org development community
+
+Asterisk is a trademark of Sangoma Technologies Corporation
+