George Joseph 4d40b161c3 stringfields: Refactor to allow fields to be added to the end of structures
String fields are great, except that you can't add new ones without breaking
ABI compatibility because it shifts down everything else in the structure.
The only alternative is to add your own char * field to the end of the
structure and manage the memory yourself which isn't ideal, especially since
you then can't use the OPT_STRINGFIELD_T type.

Background:

The reason string fields had to be declared inside the
AST_DECLARE_STRING_FIELDS block was to facilitate iteration over all declared
fields for initialization, compare and copy.  Since AST_DECLARE_STRING_FIELDS
declared the pool, then the fields, then the manager, you could use the offsets
of the pool and manager and iterate over the sequential addresses in between to
access the fields. The actual pool, field allocation and field set operations
don't actually care where the field is.  It's just iteration over the fields
that was the problem.

Solution: Extended String Fields

An extended string field is one that is declared outside the
AST_DECLARE_STRING_FIELDS block but still (anywhere) inside the parent
structure.  Other than using AST_STRING_FIELD_EXTENDED instead of
AST_STRING_FIELD, it looks the same as other string fields.  It's storage comes
from the pool and it participates in string field compare and copy operations
peformed on the parent structure. It's also a valid target for the
OPT_STRINGFIELD_T aco option type.

Implementation:

To keep track of the extended fields and make sure that ABI isn't broken, the
existing embedded_pool pointer in the manager structure was repurposed to be a
pointer to a separate header structure that contains the embedded_pool pointer
plus a vector of fields.  The length of the manager structure didn't change and
the embedded_pool pointer isn't used in the macros, only the stringfields C
code.  A side benefit of this is that changing the header structure in the
future won't break ABI.

ast_string_fields_init initializes the normal string fields and appends them to
the vector, and subsequent calls to ast_string_field_init_extended initialize
and append the extended fields. Cleanup, ast_string_fields_cmp, and
ast_string_fields_copy can now work on the vector instead of sequentially
traversing the addresses between the pool and manager.

The total size of a structure using string fields didn't change, whether using
extended fields or not, nor have the offsets of any structure members, either
inside the original block or outside.  Adding an extended field to the end of a
structure is the same as adding a char *.

Details:

The stringfield C code was pulled out from utils.c and into stringfields.c.
It just made sense.

Additional work was done in ast_string_field_init and
ast_calloc_with_stringfields to handle the allocation of the new header
structure and the vector, and the associated cleanup.  In the process some
additional NULL pointer checking was added.

A lot of work was done in stringfields.h since the logic for compare and copy
is there.  Documentation was added as well as somne additional NULL checking.

The ability to call ast_calloc_with_stringfields with a number of structures
greater than 1 never really worked.  Well, the calloc worked but there was no
way to access the additional structures or clean them up.  It was agreed that
there was no use case for requesting more than 1 structure so an ast_assert
was added to prevent it and the iteration code removed.

Testing:

The stringfield unit tests were updated to test both normal and extended
fields.  Tests for ast_string_field_ptr_set_by_fields and
ast_calloc_with_stringfields were also added.

As an ABI test, 13 was compiled from git and the res_pjsip_* modules, except
res_pjsip itself, saved off.  The patch was then added and a full compile and
install was performed.  Then the older res_pjsip_* moduled were copied over the
installed versions so res_pjsip was new and the rest were old.  No issues.

contact->aor, which is a char * at the end of contact, was then changed to an
extended string field and a recompile and reinstall was performed, again
leaving stock versions of the the res_pjsip_* modules.  Again, no issues with
the res_pjsip_* modules using the old stringfield implementation and with
contact->aor as a char *, and res_pjsip itself using the new stringfield
implementation and contact->aor being an extended string field.

Finally, several existing string fields were converted to extended string
fields to test OPT_STRINGFIELD_T.  Again, no issues.

Change-Id: I235db338c5b178f5a13b7946afbaa5d4a0f91d61
2016-04-04 19:07:53 -05:00
2015-04-11 19:43:43 -06:00
2016-01-06 09:48:35 +08:00
2012-10-13 15:14:51 +00:00
2015-06-05 11:23:16 -05:00
2015-04-11 19:43:43 -06:00
2014-07-18 00:11:37 +00:00
2016-01-20 10:52:45 -03:00

===============================================================================
===                     The Asterisk(R) Open Source PBX
===
===                   by Mark Spencer <markster@digium.com>
===                  and the Asterisk.org developer community
===
===                    Copyright (C) 2001-2016 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.

  If you downloaded Asterisk as a tarball, see the security section in the PDF
version of the documentation in doc/tex/asterisk.pdf.  Alternatively, pull up
the HTML version of the documentation in doc/tex/asterisk/index.html.  The
source for the security document is available in doc/tex/security.tex.
-------------------------------------------------------------------------------

-------------------------------------------------------------------------------
--- 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 at:

           http://www.asterisk.org

  The official Asterisk wiki can be found at:

           https://wiki.asterisk.org

  In addition you'll find lots of information compiled by the Asterisk
community on this Wiki:

           http://www.voip-info.org/wiki-Asterisk

  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 http://www.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 (www.digium.com)
	* 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
3.0 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 README 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. They are all in the configs/ directory.

2) Run "./configure"

  Execute the configure script to guess values for system-dependent
variables used during compilation.

3) 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.

4) Run "make"

  Assuming the build completes successfully:

5) 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:

6) "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 <command>".  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 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 (http://www.ntp.org/) 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.

  If this release of Asterisk was downloaded from a tarball, then some
additional documentation should have been included.
     * doc/tex/asterisk.pdf --- PDF version of the documentation
     * doc/tex/asterisk/index.html --- HTML version of the documentation

  Finally, you may wish to visit the web site and join the mailing list if
you're interested in getting more information.

   http://www.asterisk.org/support

  Welcome to the growing worldwide community of Asterisk users!
-------------------------------------------------------------------------------

--- Mark Spencer, and the Asterisk.org development community

-------------------------------------------------------------------------------
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