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From the review board request: The fix from review 298 has exposed a new bug in chan_sip. When we hang up an outgoing call, we first will dump all the outstanding packets on the sip_pvt using __sip_pretend_ack. Then, if we can, we send a CANCEL. The problem with this is that since destroyed all the outstanding packets on the dialog, we cannot match the incoming 487 response to our INVITE. Because we cannot match the response, we do not send an ACK. To correct this, instead of using __sip_pretend_ack, I have changed the code to loop through the list of packets and call __sip_semi_ack on each one instead. This causes us to stop retransmitting the requests, but we still have them around in case we get responses for them. When discussing this earlier today with Josh Colp, we both agreed that in the majority of cases, this would be enough of a fix. However, we also agreed that we should have a safety net in place in case we never receive a response to our initial INVITE. To handle this, I have modified __sip_autodestruct to behave similar to the way it does in Asterisk 1.4. If there are outstanding packets on the sip_pvt, the needdestroy flag is not set, and the last request on the dialog was either a CANCEL or BYE, then we set the needdestroy flag and reschedule destruction for 10 seconds in the future. If, though, the needdestroy flag is set, then we use __sip_pretend_ack to kill the remaining outstanding packets so that the monitor thread can destroy the sip_pvt. I ran two separate tests. First, I placed a call from my Aastra phone to my Polycom phone. I hung up the Aastra before the Polycom answered. I verified through sip debug output that Asterisk properly ACKed the 487 received from the Polycom. For my second test, I set up a SIPp UAS scenario so that it would send a 200 OK in response to a CANCEL but would not send a 487 for the original INVITE. I verified that after about 40 seconds, Asterisk properly cleans up the outgoing sip_pvt for the call. Review: https://reviewboard.asterisk.org/r/308 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Asterisk(R) Open Source PBX
by Mark Spencer <markster@digium.com>
and the Asterisk.org developer community
Copyright (C) 2001-2009 Digium, Inc.
and other copyright holders.
================================================================
* SECURITY
It is imperative that you read and fully understand the contents of
the security information file (doc/security.txt) before you attempt
to configure and run an Asterisk server.
* 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. For more information
on the project itself, please visit the Asterisk home page at:
http://www.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, 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 soundcard) to install and run Asterisk.
Supported telephony hardware includes:
* All Wildcard (tm) products from Digium (www.digium.com)
* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
* any full duplex sound card supported by ALSA or OSS
* any ISDN card supported by mISDN on Linux (BRI)
* The Xorcom AstriBank channel bank
* VoiceTronix OpenLine products
The are several drivers for ISDN BRI cards available from third party sources.
Check the voip-info.org wiki for more information on chan_capi and
zaphfc.
* 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.
To discover the major new features of Asterisk 1.2, please visit
http://edvina.net/asterisk1-2/
* 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 OpenSSL,
ncurses and zlib.
On many distributions, these files 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 modules dependencies.
4) Run "make"
Assuming the build completes successfully:
5) Run "make install"
Each time you update or checkout from the repository, you are strongly
encouraged to ensure all previous object files are removed to avoid internal
inconsistency in Asterisk. Normally, this is automatically done with
the presence of the file .cleancount, which increments each time a 'make clean'
is required, and the file .lastclean, which contains the last .cleancount used.
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 config files you have.
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 "help" at any time to get help with the system. For help
with a specific command, type "help <command>". To start the PBX using
your sound card, you can type "dial" to dial the PBX. Then you can use
"answer", "hangup", and "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'll 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 (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 approxiately 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 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
----
Asterisk is a trademark belonging to Digium, inc