7c0b8f0701
RFC7616 and RFC8760 allow more than one WWW-Authenticate or Proxy-Authenticate header per realm, each with different digest algorithms (including new ones like SHA-256 and SHA-512-256). Thankfully however a UAS can NOT send back multiple Authenticate headers for the same realm with the same digest algorithm. The UAS is also supposed to send the headers in order of preference with the first one being the most preferred. We're supposed to send an Authorization header for the first one we encounter for a realm that we can support. The UAS can also send multiple realms, especially when it's a proxy that has forked the request in which case the proxy will aggregate all of the Authenticate headers and then send them all back to the UAC. It doesn't stop there though... Each realm can require a different username from the others. There's also nothing preventing each digest algorithm from having a unique password although I'm not sure if that adds any benefit. So now... For each Authenticate header we encounter, we have to determine if we support the digest algorithm and, if not, just skip the header. We then have to find an auth object that matches the realm AND the digest algorithm or find a wildcard object that matches the digest algorithm. If we find one, we add it to the results vector and read the next Authenticate header. If the next header is for the same realm AND we already added an auth object for that realm, we skip the header. Otherwise we repeat the process for the next header. In the end, we'll have accumulated a list of credentials we can pass to pjproject that it can use to add Authentication headers to a request. NOTE: Neither we nor pjproject can currently handle digest algorithms other than MD5. We don't even have a place for it in the ast_sip_auth object. For this reason, we just skip processing any Authenticate header that's not MD5. When we support the others, we'll move the check into the loop that searches the objects. Changes: * Added a new API ast_sip_retrieve_auths_vector() that takes in a vector of auth ids (usually supplied on a call to ast_sip_create_request_with_auth()) and populates another vector with the actual objects. * Refactored res_pjsip_outbound_authenticator_digest to handle multiple Authenticate headers and set the stage for handling additional digest algorithms. * Added a pjproject patch that allows them to ignore digest algorithms they don't support. This patch has already been merged upstream. * Updated documentation for auth objects in the XML and in pjsip.conf.sample. * Although res_pjsip_authenticator_digest isn't affected by this change, some debugging and a testsuite AMI event was added to facilitate testing. Discovered during OpenSIPit 2021. ASTERISK-29397 Change-Id: I3aef5ce4fe1d27e48d61268520f284d15d650281
The Asterisk(R) Open Source PBX
By Mark Spencer <markster@digium.com> and the Asterisk.org developer community.
Copyright (C) 2001-2021 Sangoma Technologies Corporation and other copyright holders.
SECURITY
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.
WHAT IS ASTERISK ?
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 wiki. 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.
SUPPORTED OPERATING SYSTEMS
Linux
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.
Others
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.
GETTING STARTED
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 (http://www.quicknet.net)
- 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
UPGRADING FROM AN EARLIER VERSION
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.
NEW INSTALLATIONS
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:
- 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.
- Run
./configure
Execute the configure script to guess values for system-dependent variables used during compilation.
- Run
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.
- Run
make
Assuming the build completes successfully:
- Run
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:
- Run
make samples
Doing so will overwrite any existing configuration files you have installed.
- Finally, you can launch Asterisk in the foreground mode (not a daemon) with:
# 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 ". To start the PBX using your sound card, you can type "console dial" to dial the PBX. Then you can use "console answer", "console hangup", and "console dial" to simulate the actions of a telephone. Remember that if you don't have a full duplex sound card (and Asterisk will tell you somewhere in its verbose messages if you do/don't) then it won't work right (not yet).
"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.
ABOUT CONFIGURATION FILES
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.
SPECIAL NOTE ON TIME
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.
FILE DESCRIPTORS
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:
PAM-BASED LINUX SYSTEM
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.
GENERIC UNIX SYSTEM
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.
MORE INFORMATION
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