This patch does the following:
1) The env scripts have been updated to be tolerant of a NULL configuration
file. This occurs when configuration is provided by an external script,
such that the actual config.ini file is not used.
2) Enum types have all been given names. This is needed for PostgreSQL script
generation.
3) The identifier meetme_confno_starttime_endtime is greater than 30
characters, and hence invalid for Oracle databases. This has been truncated
down to meetme_confno_start_end.
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The patch from ASTERISK-21965 was committed perhaps a bit too hastily. Walter
and Tzafrir have pointed out numerous issues with the approach and have
propsed an alternative in r/2757. Since it's not a time critical issue and
is not worth holding up the release of 12 for it, I've gone ahead and reverted
r394939 from 12/trunk and re-opened ASTERISK-21965.
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This patch replaces contrib/realtime/ with a new setup for managing the
database schema required for database integration with Asterisk. In
addition to initializing a database with the proper schema, alembic can do a
database migration to assist with upgrading Asterisk in the future.
Hopefully this helps make setting up and operating Asterisk with a database
easier.
With this the schema only needs to be maintained in one place instead of
once per database. The schemas I have added here have a bit of improvement
over the examples that were there before (some added consistency and added
some missing indexes). Managing the schema in one place here also applies
to all databases supported by SQLAlchemy.
See contrib/ast-db-manage/README.md for more details.
Review: https://reviewboard.asterisk.org/r/2731
patch by Russell Bryant (license 6300)
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Most, if not all, of the backing features of a conf file should now be
implemented (e.g. multi-line comments, includes, templates, etc...). A
few of the options still need to be mapped. Those are currently listed
in the 'sip_to_res_sip.py' file.
Things to do:
(1) There is more work to do here, at least for the sip.conf items that
aren't currently parsed. An issue will be created for that.
(2) All of the scripts should probably be passed through pylint and have
as many PEP8 issues fixed as possible.
(3) A public review is probably warranted at that point of the entire script.
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents XML documentation duplication by expanding channel and
bridge snapshot tags into channel and bridge snapshot parameter sets
with a given prefix or defaulting to no prefix. This also prevents
documentation from becoming fractured and out of date by keeping all
variations of the documentation in template form such that it only
needs to be updated once and keeps maintenance to a minimum.
Review: https://reviewboard.asterisk.org/r/2708/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch modifies the behavior of safe_asterisk in two ways:
(1) It modifies the Asterisk Makefile such that safe_asterisk is always
installed on a 'make install'. This was done as bugfixes in the
safe_asterisk script were not applied in previous version of Asterisk
without first removing the old version of the script.
(2) In order to keep a newly installed version of safe_asterisk from impacting
local modifications, a new config file - safe_asterisk.conf.sample - has
been provided. Settings that were previously modified in safe_asterisk can
be set there instead.
(closes issue ASTERISK-21965)
Reported by: Jeremy Kister
patches:
safe_asterisk.patch uploaded by jkister (License 6232)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
** This script is in no way finished.
Started the initial "cut" at converting a sip.conf file to a res_sip.conf file.
Hopefully the bulk of the framework is in place and only a few minor adjustments
need to be made when an option mapping is added that "doesn't fit". This script
and supporting files should be executable against python version 2.5.
An OrderedDict class (backported from a newer version of python) is included.
A MultiOrderedDict class is implemented so options, when added, should be able
to be added in order and allowed to have multiple values.
Currently the scripts supports the majority of endpoint options found in
res_sip.conf. Support has also been added for Aor(s) and the ACL/security
sections. Inside the sip_to_res_sip.py file one can see a list of options
that still need to be mapped.
Also items that still need to be done: templates, includes, parsing '=>'
delimiter. Note that some code is hopefully in place already to support
templates (e.g. lookup/retrieving defaults from them). However, the
parsing of and adding of the section needs to be done.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Don't regenrate cat.cfg, ca.crt and ca.key if they were already created
on a previous run.
(closes issue ASTERISK-21932)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393383 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch updates the autosupport script to collect all information available
to the Asterisk CLI command "digium_phones". It also makes minor improvements
in options handling.
(closes issue AST-1163)
Reported by: Trey Blancher
patches:
390347_autosupport.diff uploaded by tblancher (License 5821)
390348_autosupport.diff uploaded by tblancher (License 5821)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When r383579 was committed, it made Jansson a required dependency.
While libjansson-dev and jansson-devel are available on recent
distros, some older (but still supported) distros don't have
it. There's a pull request[1] to get it into repoforge, but that still
doesn't help everyone. (And helps no one until the pull request is
merged and packages are built).
This patch adds Jansson install from source to the install_unpackaged()
function. There are a few gotcha's, which makes this change not
completely trivial.
* Since Jansson may be installed by a package, don't install from
source if a package installation can be found
* libresample may also be installed via package, so I added a
similar check to that.
* Since Jansson installs into /usr/local, this patch also adds
/usr/local/lib to /etc/ld.so.conf.d so that the library can be
found.
* The alternative was to install into /usr, but then it gets
complicated having to deal with EL's /usr/lib{32,64} shenanigans.
[1]: https://github.com/repoforge/rpms/pull/250
Review: https://reviewboard.asterisk.org/r/2414/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds support for RFC 3327 "Path" headers. This can be enabled in
sip.conf using the 'supportpath' setting, either on a global basis or on a
peer basis. This setting enables Asterisk to route outgoing out-of-dialog
requests via a set of proxies by using a pre-loaded route-set defined by the
Path headers in the REGISTER request. This patch also adds Realtime support
for dynamically updating the Path information for a peer.
A huge thank-you to Klaus Darillion and Olle E Johansson for their efforts
in writing this patch.
Review: https://reviewboard.asterisk.org/r/2235/
Review: https://reviewboard.asterisk.org/r/991/
(closes issue ASTERISK-16884)
Reported by: klaus3000
Tested by: klaus3000, oej, mjordan
patches:
path-1.8.0-patch.txt uploaded by klaus3000 (License 5054)
oolong-path-support-trunk in team branch by oej (License 5267)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The original report had to do with a realtime peer behind NAT being pruned and
the peer's private address being used instead of its external address. Upon
debugging, it was discovered that this was being caused by the addition of
the auto_force_rport and auto_comedia settings.
This patch does the following:
* Adds a missing note to the CHANGES file indicating that the default global nat
setting is auto_force_rport
* Constify the 'req' parameter for check_via()
* Add calls to check_via() in a couple of places in order for the auto_*
settings to do their job in attempting to determine if NAT is involved
* Set the flags SIP_NAT_FORCE_RPORT and SIP_PAGE2_SYMMETRICRTP if the auto_*
settings are in use where it was needed
* Moves the copying of peer flags up in build_peer() to before they are used;
this fixes the realtime prune issue
* Update the contrib/realtime schemas to allow the nat column to handle the
different nat setting combinations we have
This patch received a review and "Ship It!" on the issue itself.
(closes issue ASTERISK-20904)
Reported by: JoshE
Tested by: JoshE, Michael L. Young
Patches:
asterisk-20904-nat-auto-and-rt-peersv2.diff Michael L. Young (license 5026)
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This adds the ability to get the DPMA version, a listing of the local
firmware directory, and indexes of configured remote directories.
(closes issue AST-1070)
Reported By: Malcolm Davenport
Tested By: Kinsey Moore <kmoore@digium.com>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380004 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When r376428 was commited to re-order start up sequences to be more tolerant of
forking with thread primitives, a few items were changed that caused changes
in behavior on some distros. This includes:
* Not displaying the splash screen on a remote console.
* Displaying an error message on stderr when a remote console cannot connect
to a running instance of Asterisk.
In the first case, the splash screen was re-added (thanks to Michael L. Young).
In the second case, the various init.d scripts were modified to pipe stderr
to /dev/null, as the error message is useful - if you execute a remote
console or a remote console command execution and it fail, it should tell
you. Note that the error message was always present, it just failed to be
printed prior to r376428.
Much thanks to the folks who quickly reported this problem, provided solutions,
and promptly tested the various init.d scripts on a variety of distros.
(closes issue ASTERISK-20945)
Reported by: Warren Selby
Tested by: Michael L. Young, Jamuel Starkey, kaldemar, Danny Nicholas, mjordan
patches:
asterisk-20945-remote-intro-msg.diff uploaded by elguero (license 5026)
ASTERISK-20945-1.8-mjordan.diff uploaded by mjordan (license 6283)
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This provides a JSON API by pulling in and wrapping the Jansson JSON
library[1]. The Asterisk API basically mirrors the Jansson
functionality, with a few minor tweaks.
* Some names have been asteriskified to protect the innocent.
* Jansson provides both reference-stealing and reference-borrowing
versions of several API's. The Asterisk API is exclusively
reference-stealing for operations that put elements into arrays and
objects.
* No support for doubles, since we usually don't need that.
* Coming along for the ride is the ast_test_validate macro, which made
the unit tests much easier to write.
[1]: http://www.digip.org/jansson/
(issue ASTERISK-20887)
(closes issue ASTERISK-20888)
Review: https://reviewboard.asterisk.org/r/2264/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is an interesting feature that allows additional strings to be used to
search the Directory, primarily intended to be used with nicknames, but could
be used with affiliations and the like. Because the name field is used in
more than one place (such as email notifications), it is important that these
additional strings not be placed in the name field, but be specified
separately.
Review: https://reviewboard.asterisk.org/r/2244/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In ASTERISK-20726 UUID was added to Asterisk. This commit is to add the dependancies to the install script
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Update and extend the configuration_file group and enable linking. Commit other cleanups from multi-version Doxygen testing. Update title that was left behind many years ago.
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
During testing I used an alternate output directory and mistakenly committed it. Matt Jordan noticed and I reverted. This is the correct setting for local output to match with all branches.
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add Doxygen to the Debian install list. I will check for other platforms like Red Hat
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Revert a local testing config that I made. This was not intended to be committed.
Thank you Matt Jordan for noticing this.
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Whitespace, doc-blocks, spelling, case, missing and incorrect tags.
* Add cleanup to Makefile for the Doxygen configuration update
* Start updating Doxygen configuration for cleaner output
* Enable inclusion of configuration files into documentation
* remove mantisworkflow...
* update documentation README
* Add markup to Tilghman's email and talk with him about updating his email, he knows...
* no code changes on this commit other than the mentioned Makefile change
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This annoying update is almost totally whitespace and updated config comments. I did add Python to the documented file types.
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
All voicemails now have a 'msg_id' included in their metadata. The ODBC
message storage backend now requires this column; as such, the MySQL contrib
script that creates the voicemail_data table has been updated with the appropriate
column information.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch changes the Asterisk configure script and build system to detect
the presence of the NetBSD editline library (libedit) on the system. If it is
found, it will be used in preference to the version included in the Asterisk
source tree.
(closes issue ASTERISK-18725)
Reported by: Jeffrey C. Ollie
Review: https://reviewboard.asterisk.org/r/1528/
Patches:
0001-Allow-linking-building-against-an-external-editline.patch uploaded by jcollie (license #5373) (heavily modified by kpfleming)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
contrib/scripts/live_ast currently assumes that it is being run from the
top-level directory of the source tree. It creates a script that will
also set the working directory.
This fix avoids the need to set the working directory if the caller sets
LIVE_AST_BASE_DIR instead.
It relies on realpath for that. If realpath is not available, it will
fall back to the original behaviour.
Review: https://reviewboard.asterisk.org/r/2027/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Presence support has been added. This is accomplished by
allowing for presence hints in addition to device state
hints. A dialplan function called PRESENCE_STATE has been
added to allow for setting and reading presence. Presence
can be transmitted to Digium phones using custom XML
elements in a PIDF presence document.
Voicemail has new APIs that allow for moving, removing,
forwarding, and playing messages. Messages have had a new
unique message ID added to them so that the APIs will work
reliably. The state of a voicemail mailbox can be obtained
using an API that allows one to get a snapshot of the mailbox.
A voicemail Dialplan App called VoiceMailPlayMsg has been
added to be able to play back a specific message.
Configuration hooks have been added. Configuration hooks
allow for a piece of code to be executed when a specific
configuration file is loaded by a specific module. This is
useful for modules that are dependent on the configuration
of other modules.
chan_sip now has a public method that allows for a custom
SIP INFO request to be sent mid-dialog. Digium phones use
this in order to display progress bars when files are played.
Messaging support has been expanded a bit. The main
visible difference is the addition of an AMI action
MessageSend.
Finally, a ParkingLots manager action has been added in order
to get a list of parking lots.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added ability to use multiple lines on phone, so for one device in configuration multiple lines can be defined, it allows to have multiple calls on one phone, callwaiting and switching between calls.
* Added ability for translation on-screen menu to multiple languages. Tested on Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4, ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen menu of phone
* Other described in CHANGES file
Testing done by issue tracker users: ibercom, scsiborg, idarwin, TeknoJuce, c0rnoTa.
Tested on production system by Jonn Taylor (jonnt) using phone models: Nortel i2004, 1120E and 1140E.
(closes issue ASTERISK-16890)
Review: https://reviewboard.asterisk.org/r/1243/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
actually remove res_ais. This commit removes it.
In addition, the 'install_prereq' script has been updated to no longer install
AIS dependency packages, and instead install Corosync packages instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354459 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is based on the one by David Vossel, developer extrodinaire, at
https://reviewboard.asterisk.org/r/344/. If multiple peers are defined with the
same host/port, but differing callbackextensions, it chooses the peer with the
matching callbackextension. Since callbackextension creates an outbound
registration with the callbackextension as the Contact address, matching an
incoming request by that (in addition to the host/port) makes a lot of sense.
This patch also adds support for callbackextension to realtime by querying all
peers with callbackextensions on reload and adding registrations for them.
(closes issue ASTERISK-13456)
Review: https://reviewboard.asterisk.org/r/344/
Review: https://reviewboard.asterisk.org/r/1717/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
be empty, don't check for it, instead of check if LD_LIBRARY_PATH is
already set and if so, append LIVE_AST_LD_PATH_EXTRA properly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r350788 | kpfleming | 2012-01-14 09:22:33 -0600 (Sat, 14 Jan 2012) | 8 lines
Ensure that two prerequisites are properly installed on Debian-style distributions.
* Don't specify a specific version of libgmime; newer versions are available
now and acceptable.
* Install libsrtp so that res_srtp can be built.
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r350789 | kpfleming | 2012-01-14 09:23:32 -0600 (Sat, 14 Jan 2012) | 3 lines
Correct some 'set-but-not-used' variable warnings.
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r350127 | rmudgett | 2012-01-09 12:40:33 -0600 (Mon, 09 Jan 2012) | 12 lines
Update contrib script live_ast to invoke Asterisk with valgrind and suppression file.
* Added valgrind_compare script to compare two valgrind log files for
differences.
(issue ASTERISK-17339)
Reported by: Tzafrir Cohen
Patches:
valgrind_compare (license #5035) script uploaded by Tzafrir Cohen
live_ast_valgrind.diff (license #5035) patch uploaded by Tzafrir Cohen
live_ast_valgrind_v2.diff (license #5185) patch uploaded by Paul Belanger
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r350128 | rmudgett | 2012-01-09 12:54:56 -0600 (Mon, 09 Jan 2012) | 11 lines
live_ast: valgrind: run asterisk under valgrind
Adds a new sub-command, "valgrind" to live_ast. It runs asterisk under
valgrind. The extra command-line parameters are passed to Asterisk as
usual, and parameters to valgrind are passed through LIVE_AST_VALGRIND_ARGS
in live.conf .
Review: https://reviewboard.asterisk.org/r/1109/
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Added information collection from the output of the utilities: top, free, uptime, ifconfig
Added information collection from the output of the Asterisk command 'dahdi show status'
Added option / flag '-n, --non-interactive'
Updated man page to reflect new option / flag '-n, --non-interactive'
Patch-by: John Bigelow (itzanger)
(closes issue AST-749)
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This patch allows the imapserver, imapport, and imapflags settings to be
overridden for any voicemail user. It also documents the settings in
the sample voicemail.conf file, and updates the voicemail schema to
allow storage of those columns.
(closes issue ASTERISK-16489)
Reporter: Hubert Mickael
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1614/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
environment variables and also enables a custom run directory for asterisk
(instead of defaulting to /tmp).
Patch by: Byron Clark (byronclark)
(closes ASTERISK-17810)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds two new menu features to app_confbridge, admin_toggle_menu_
participants and participant_count. The admin action will globally mute /
unmute all non-admin participants on a converence, while the participant
count simply exposes the existing participant count function to the
conference bridge menu.
This also adds configuration options to change the sound played when the
conference is globally muted / unmuted, as well as the necessary config
hooks to place these functions in the DTMF menus.
(closes issue ASTERISK-18204)
Reported by: Kevin Reeves
Tested by: Matt Jordan
Patches:
app_confbridge.c.patch.txt, conf_config_parser.c.patch.txt,
confbridge.h.patch.txt uploaded by Kevin Reeves (license 6281)
Review: https://reviewboard.asterisk.org/r/1518/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This option is not only useless, but has been broken since inception since
the flag was never copied from the peer where it is set to the pvt where
it was checked. RFC 3261 specificially states that you should not send a
provisional response to a non-INVITE request, and if we did fix the code
so that it worked, it would cause the same kind of user enumeration
vulnerability that we've discussed with the nat= setting. This patch
removes registertrying option and any code that would have sent a 100
response to a register.
Review: https://reviewboard.asterisk.org/r/1562/
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r337115 | lmadsen | 2011-09-20 17:18:25 -0500 (Tue, 20 Sep 2011) | 7 lines
Update RedHat Init script to work with Heartbeat.
The current RedHat init script was not LSB compatible. This change will make it LSB compatible so that
it can work correctly with Heartbeat.
(Closes issue ASTERISK-18253)
Reported by: c0rnoTa
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This permits the list of codecs to be specified in one configuration line,
instead of two or more, generally with the aim of either allowing all codecs
with the exception of a few or disallowing most but permitting a few.
Review: https://reviewboard.asterisk.org/r/1411/
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Adds a new sub-command, "valgrind" to live_ast. It runs asterisk under
valgrind. The extra command-line parameters are passed to Asterisk as
usual, and parameters to valgrind are passed through LIVE_AST_VALGRIND_ARGS
in live.conf .
Review: https://reviewboard.asterisk.org/r/1109/
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Several variables in the script control which files are converted and the
source and destination formats.
Patch-by: Trey Blancher <support@digium.com>
(closes AST-560)
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r307536 | qwell | 2011-02-10 16:39:30 -0600 (Thu, 10 Feb 2011) | 22 lines
Merged revisions 307535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r307535 | qwell | 2011-02-10 16:35:49 -0600 (Thu, 10 Feb 2011) | 15 lines
Merged revisions 307534 via svnmerge from
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r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) | 8 lines
Remove color when executing commands via a remote console.
Essentially this makes '-x' imply '-n' on rasterisk. This was done in a
different and incomplete way previously, which I'm reverting here.
(issue #18776)
Reported by: alecdavis
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r292787 | lmadsen | 2010-10-22 16:28:43 -0500 (Fri, 22 Oct 2010) | 21 lines
Merged revisions 292786 via svnmerge from
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r292786 | lmadsen | 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010) | 13 lines
Update the LDIF file for LDAP.
The LDIF file asterisk.ldif was quite a bit out of date from the asterisk.ldap-schema file, so I've
now updated that to be in sync. The asterisk.ldif file being out of sync was a problem on my systems
where I was doing an ldapadd to import the schema into the LDAP database, and the existing file
would cause problems and ERROR messages when registering.
Additional documention has been added based on feedback in the issue I'm closing.
(closes issue #13861)
Reported by: scramatte
Patches:
ldap-update.txt uploaded by lmadsen (license 10)
Tested by: lmadsen, jcovert, suretec, rgenthner
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r292740 | twilson | 2010-10-22 09:49:34 -0700 (Fri, 22 Oct 2010) | 45 lines
Add TLS cert helper script
This script is useful for quickly generating self-signed CA, server, and client
certificates for use with Asterisk. It is still recommended to obtain
certificates from a recognized Certificate Authority and to develop an
understanding how SSL certificates work. Real security is hard work.
OPTIONS:
-h Show this message
-m Type of cert "client" or "server". Defaults to server.
-f Config filename (openssl config file format)
-c CA cert filename (creates new CA cert/key as ca.crt/ca.key if not passed)
-k CA key filename
-C Common name (cert field)
For a server cert, this should be the same address that clients
attempt to connect to. Usually this will be the Fully Qualified
Domain Name, but might be the IP of the server. For a CA or client
cert, it is merely informational. Make sure your certs have unique
common names.
-O Org name (cert field)
An informational string (company name)
-o Output filename base (defaults to asterisk)
-d Output directory (defaults to the current directory)
Example:
To create a CA and a server (pbx.mycompany.com) cert with output in /tmp:
ast_tls_cert -C pbx.mycompany.com -O "My Company" -d /tmp
This will create a CA cert and key as well as asterisk.pem and the the two
files that it is made from: asterisk.crt and asterisk.key. Copy asterisk.pem
and ca.crt somewhere (like /etc/asterisk) and set tlscertfile=/etc/asterisk.pem
and tlscafile=/etc/ca.crt. Since this is a self-signed key, many devices will
require you to import the ca.crt file as a trusted cert.
To create a client cert using the CA cert created by the example above:
ast_tls_cert -m client -c /tmp/ca.crt -k /tmp/ca.key -C "Joe User" -O \
"My Company" -d /tmp -o joe_user
This will create client.crt/key/pem in /tmp. Use this if your device supports
a client certificate. Make sure that you have the ca.crt file set up as
a tlscafile in the necessary Asterisk configs. Make backups of all .key files
in case you need them later.
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Kevin Harwell
Mark Michelson
Matthew Jordan
David M. Lee
Jonathan Rose
Kinsey Moore
Jason Parker
Tzafrir Cohen
Michael L. Young
Andrew Latham
Tilghman Lesher
Richard Mudgett
Russell Bryant
Kevin P. Fleming
Igor Goncharovskiy
Terry Wilson
Sean Bright
Stefan Schmidt
Matthew Nicholson
Walter Doekes
Gregory Nietsky
Leif Madsen
Paul Belanger
David Ruggles