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4bdf5d329f
This patch fixes 2 original issues and more that those 2 exposed. * When we send a NOTIFY, and the client either doesn't respond or responds with a non OK, pjproject only calls our pubsub_on_evsub_state callback, no others. Since pubsub_on_evsub_state (which does the sub_tree cleanup) does not expect to be called back without the other callbacks being called first, it just returns leaving the sub_tree orphaned. Now pubsub_on_evsub_state checks the event for PJSIP_EVENT_TSX_STATE which is what pjproject will set to tell us that it was the transaction that timed out or failed and not the subscription itself timing our or being terminated by the client. If is TSX_STATE, pubsub_on_evsub_state now does the proper cleanup regardless of the state of the subscription. * When a client renews a subscription, we don't update the persisted subscription with the new expires timestamp. This causes subscription_persistence_recreate to prune the subscription if/when asterisk restarts. Now, pubsub_on_rx_refresh calls subscription_persistence_update to apply the new expires timestamp. This exposed other issues however... * When creating a dialog from rdata (which sub_persistence_recreate does from the packet buffer) there must NOT be a tag on the To header (which there will be when a client refreshes a subscription). If there is one, pjsip_dlg_create_uas will fail. To address this, subscription_persistence_update now accepts a flag that indicates that the original packet buffer must not be updated. New subscribes don't set the flag and renews do. This makes sure that when the rdata is recreated on asterisk startup, it's done from the original subscribe packet which won't have the tag on To. * When creating a dialog from rdata, we were setting the dialog's remote (SUBSCRIBE) cseq to be the same as the local (NOTIFY) cseq. When the client tried to resubscribe after a restart with the correct cseq, we'd reject the request with an Invalid CSeq error. * The acts of creating a dialog and evsub by themselves when recreating a subscription does NOT restart pjproject's subscription timer. The result was that even if we did correctly recreate the subscription, we never removed it if the client happened to go away or send a non-OK response to a NOTIFY. However, there is no pjproject function exposed to just set the timer on an evsub that wasn't created by an incoming subscribe request. To address this, we create our own timer using ast_sip_schedule_task. This timer is used only for re-establishing subscriptions after a restart. An earlier approach was to add support for setting pjproject's timer (via a pjproject patch) and while that patch is still included here, we don't use that call at the moment. While addressing these issues, additional debugging was added and some existing messages made more useful. A few formatting changes were also made to 'pjsip show scheduled tasks' to make displaying the subscription timers a little more friendly. ASTERISK-26696 ASTERISK-26756 Change-Id: I8c605fc1e3923f466a74db087d5ab6f90abce68e
===============================================================================
=== 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
-------------------------------------------------------------------------------
Asterisk is a trademark of Digium, Inc.