After the nightly update of the documentation on asterisk.org, I'll post
an update to asterisk-dev with a pointer to the changes. This covers some
release branch and commit policy information. None of this should be a
surprise, since it's just documenting what we have already been doing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162418 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008) | 8 lines
Remove the test_for_thread_safety() function completely.
The test is not valid. Besides, if we actually suspected that recursive
mutexes were not working, we would get a ton of LOG_ERROR messages when
DEBUG_THREADS is turned on.
(inspired by a discussion on the asterisk-dev list)
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Based on cli_permissions.conf configuration file, we are able to permit or deny
cli commands based on some patterns and the local user and group running rasterisk.
(Sorry if I missed some of the testers).
Reviewboard: http://reviewboard.digium.com/r/11/
(closes issue #11123)
Reported by: eliel
Tested by: eliel, IgorG, Laureano, otherwiseguy, mvanbaak
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) | 7 lines
Move the sanity check that makes sure "always fork" is not set along with the
console option to be after the code that reads options from asterisk.conf.
This resolves a situation where Asterisk can start taking up 100% when
misconfigured.
(Thanks to Bryce Porter (x86 on IRC) for letting me log in to his system to
figure out what was causing the 100% CPU problem.)
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- Add CLI aliases module to asterisk.
- Remove all deprecated CLI commands from the code
Initial work done by file.
Junk-Y and lmadsen did a lot of work and testing to
get the list of deprecated commands into the configuration file.
Deprecated CLI commands are now handled by this new module,
see cli_aliases.conf for more info about that.
ok russellb@ via reviewboard
(closes issue #13735)
Reported by: mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format. Currently, a new format is available for
applications and dialplan functions. A good number of conversions to the new format
are also included.
For more information, see the following message to asterisk-dev:
http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r151905 | russell | 2008-10-25 05:59:02 -0500 (Sat, 25 Oct 2008) | 8 lines
Move AMI initialization to occur after loading modules. This prevents a
deadlock when someone tries to initiate a module reload from the AMI just
as Asterisk is starting.
(closes issue #13778)
Reported by: hotsblanc
Fix suggested by hotsblanc
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read from the .asterisk_history file (and subsequently being duplicated
when written). We weren't checking the result of fgets() which meant
that we read the same line twice before feof() actually returned non-
zero.
Also, stop writing out an extra blank line between each item in the
history file, fix a minor off-by-one error, and use symbolic constants
rather than a hardcoded integer.
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when a file is invalid from when a file is missing. This is most important when
we have two configuration files. Consider the following example:
Old system:
sip.conf users.conf Old result New result
======== ========== ========== ==========
Missing Missing SIP doesn't load SIP doesn't load
Missing OK SIP doesn't load SIP doesn't load
Missing Invalid SIP doesn't load SIP doesn't load
OK Missing SIP loads SIP loads
OK OK SIP loads SIP loads
OK Invalid SIP loads incompletely SIP doesn't load
Invalid Missing SIP doesn't load SIP doesn't load
Invalid OK SIP doesn't load SIP doesn't load
Invalid Invalid SIP doesn't load SIP doesn't load
So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed. Worse yet, the old
system would do this with no indication that anything was even wrong.
(closes issue #10690)
Reported by: dtyoo
Patches:
20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
respectively. Also, take the opportunity to clean up the CLI prompt
generation code.
(closes issue #13175)
Reported by: eliel
Patches:
cliprompt.patch uploaded by eliel (license 64)
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to Asterisk licensing information. The licensing page includes the Asterisk license,
as well as a (not yet complete) list of 3rd party libraries that may be used, as well
as what license we receive them under.
Help filling out this list in the format that I have started in doxyref.h would be
much appreciated. :)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) | 12 lines
As suggested by seanbright, the PSEUDO_CHAN_LEN in
app_chanspy should be set at load time, not at compile
time, since dahdi_chan_name is determined at load time.
Also changed the next_unique_id_to_use to have the
static qualifier.
Also added the dahdi_chan_name_len variable so that
strlen(dahdi_chan_name) isn't necessary. Thanks to
seanbright for the suggestion.
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calculate the number of bytes from a sysinfo structure.
unsigned long.
(closes issue #13057)
Reported by: eliel
Patches:
asterisk.c.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131445 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines
allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places
don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it
get app_rpt building again after the DAHDI changes
(closes issue #12911)
Reported by: tzafrir
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- Convert chan_iax2 to use the timing API
- Convert usage of timing in the core to use the timing API instead of
using DAHDI directly
- Make a change to the timing API to add the set_rate() function
- change the timing core to use a rwlock
- merge a timing implementation, res_timing_dahdi
Basic testing was successful using res_timing_dahdi
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DUNDi uses a concept called the Entity ID for unique server identifiers. I have
pulled out the handling of EIDs and made it something available to all of Asterisk.
There is now a global Entity ID that can be used for other purposes as well, such
as code providing distributed device state, which is why I did this. The global
Entity ID is set automatically, just like it was done in DUNDi, but it can also be
set in asterisk.conf. DUNDi will now use this global EID unless one is specified
in dundi.conf.
The current EID for the system can be seen in the "core show settings" CLI command.
It is also available in the dialplan via the ENTITYID variable.
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and off for new installations. This includes the translation from pipes to commas
for pbx_realtime and the EXEC command for AGI, as well as the change to the Set
application not to support multiple variables at once.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r118465 | tilghman | 2008-05-27 13:58:09 -0500 (Tue, 27 May 2008) | 8 lines
NULL character should terminate only commands back to the core, not log
messages to the console.
(closes issue #12731)
Reported by: seanbright
Patches:
20080527__bug12731.diff.txt uploaded by Corydon76 (license 14)
Tested by: seanbright
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display this information in the "core show settings" CLI command. This is
useful if you want to verify that you're running a build with DONT_OPTIMIZE,
DEBUG_THREADS, etc.
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A task wraps a callback function pointer and a data pointer and is managed internal to the taskprocessor subsystem. The callback function is responsible for releasing task data.
Taskprocessor API
* ast_taskprocessor_get(..) - returns a reference to a taskprocessor
* ast_taskprocessor_unreference(..) - releases reference to a taskprocessor
* ast_taskprocessor_push(..) - push a task into a taskprocessor queue
Check doxygen for more details
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines
Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases.
(closes issue #10058)
Reported by: tracinet
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actual problems, per se. I also added format attributes to any printf wrapper functions I found that didn't have them. -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode.
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LIST instead of an RWLIST. The way this list works makes it such that
a RWLIST provides no additional benefit. Also, a mutex is needed for
use with the thread condition.
Merged revisions 105563 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008) | 24 lines
Merge in some changes from team/russell/autoservice-nochans-1.4
These changes fix up some dubious code that I came across while auditing what
happens in the autoservice thread when there are no channels currently in
autoservice.
1) Change it so that autoservice thread doesn't keep looping around calling
ast_waitfor_n() on 0 channels twice a second. Instead, use a thread condition
so that the thread properly goes to sleep and does not wake up until a
channel is put into autoservice.
This actually fixes an interesting bug, as well. If the autoservice thread
is already running (almost always is the case), then when the thread goes
from having 0 channels to have 1 channel to autoservice, that channel would
have to wait for up to 1/2 of a second to have the first frame read from it.
2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no
channels and no fds to poll() on, such as was the case with the previous code
for the autoservice thread. In this case, the code would call alloca(0), and
pass the result as the first argument to poll(). In this case, the 2nd
argument to poll() specified that there were no fds, so this invalid pointer
shouldn't actually get dereferenced, but, this code makes it explicit and
ensures the pointers are NULL unless we have valid data to put there.
(related to issue #12116)
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(closes issue #8925)
About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI
commands in Asterisk 1.4 for the next version of their book, they documented
a lot of inconsistencies. This set of changes addresses all of these issues
and has been reviewed by Leif.
While this does introduce even more changes to the CLI command structure, it
makes everything consistent, which is the most important thing.
Thanks to all that helped with this one!
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immediately at startup. Any commands in the startup_commands file in the Asterisk
config diretory will get executed.
(closes issue #11781)
Reported by: jamesgolovich
Patches:
asterisk-startupcmds.diff.txt uploaded by jamesgolovich (license 176)
-- With some changes by me.
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revision changed, every module that used the version was getting rebuilt after
every svn update. This severly annoyed me pretty quickly, so I have improved
the situation.
Now, instead of generating version.h, main/version.c is generated. version.c
includes the version information, as well as a couple of API calls for modules
to retrieve the version. So now, only version.c will get rebuilt, and the main
asterisk binary relinked, which is must faster than rebuilding http.c, manager.c,
asterisk.c, relinking the asterisk binary, chan_sip.c, func_version.c, res_agi ...
The only minor change in behavior here is that the version information reported by
chan_sip, for example, is the version of the Asterisk core, and not necessarily the
Asterisk version that the chan_sip module came from.
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so that paths and filename are writable by asterisk.c without
causing segfaults.
This involves defining the variables as const char *,
and having them point to as static, writable buffer
defined in asterisk.c
On passing, fix some errors in using these variables
in some files in utils/ , and in res/snmp/agent.c
which was redefining a variable without using paths.h
(not applicable to 1.4)
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Asterisk goes into a busy loop, the machine will be recoverable. We'd still
need to do a restart to put Asterisk back into high priority mode, but at
least a reboot won't be required. (Closes issue #11559)
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modified.
This requires casting the strings in asterisk.c when writing to
them, so we do it through a macro to do it consistently.
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globally visible.
Document the fact that DEFAULT_TMP_DIR cannot be overridden
from the default configuration (this needs to be fixed, as you
could have a totally different spooldir configured at runtime,
and yet DEFAULT_TMP_DIR keeps the compile-time default).
Remove two unused entries for sounds and images.
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build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
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- the *_CURRENT macros no longer need the list head pointer argument
- add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r85532 | russell | 2007-10-13 00:24:33 -0500 (Sat, 13 Oct 2007) | 8 lines
Properly handle the case where read() may return the text for more than one
CLI command at once for a remote console.
(closes issue #10888)
Reported by: jamesgolovich
Patches:
asterisk-climultiple.diff.txt uploaded by jamesgolovich (license 176)
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Michiel van Baak
Russell Bryant
Eliel C. Sardanons
Tilghman Lesher
Kevin P. Fleming
Sean Bright
Mark Michelson
Brett Bryant
Jeff Peeler
Dwayne M. Hubbard
Jason Parker
Joshua Colp
Terry Wilson
Luigi Rizzo
Olle Johansson