bb19e7feb5
In order to retry outbound registrations for some situations, we need access to the tdata from the original request. For instance, for 401/407 responses we need it to properly construct the subsequent request with the authentication. We also need it if we're iterating over a DNS SRV response record set so we can skip entries we've already tried. We've been getting the tdata from the server response rdata and transaction but that only works for the failures where there was actually a response (4XX, 5XX, etc). For timeouts there's no response and therefore no rdata or transaction from which to get the tdata. When processing a single A/AAAA record for a server, this wasn't an issue as we just retried that same server after the retry timer expired. If we got an SRV record set for the server though, without the state from the tdata, we just kept trying the first entry in the set repeatedly instead of skipping to the next one in the list. * Added a "last_tdata" member to the client state structure to keep track of the sent tdata. * Updated registration_client_send() to save the tdata it used into the client_state. * Updated sip_outbound_registration_response_cb() to use the tdata saved in client_state when we don't get a response from the server. We still use the tdata from the transaction when we DO get a response from the server so we can properly handle 4XX responses where our new request depends on it. General note on timeouts: Although res_pjsip_outbound_registration skips to the next record immediately when a timeout occurs during SRV set traversal, it's pjproject that determines how long to wait before a timeout is declared. As with other SIP message types, pjproject will continue trying the same server at an interval specified by "timer_t1" until "timer_b" expires. Both of those timers are set in the pjsip.conf "system" section. ASTERISK-28746 Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06
The Asterisk(R) Open Source PBX
By Mark Spencer <markster@digium.com> and the Asterisk.org developer community.
Copyright (C) 2001-2019 Digium, Inc. 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 Digium
- 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 Digium, Inc.