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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009) | 2 lines
Update the copyright year for the main page of the doxygen documentation.
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process being inherited by init.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@178342 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #14513)
Reported by: snuffy
Patches:
bug14513_fixsysinfo.diff uploaded by snuffy (license 35)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@177913 f38db490-d61c-443f-a65b-d21fe96a405b
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This patch includes a number of changes to the indications API. The primary
motivation for this work was to improve stability. The object management
in this API was significantly flawed, and a number of trivial situations could
cause crashes.
The changes included are:
1) Remove the module res_indications. This included the critical functionality
that actually loaded the indications configuration. I have seen many people
have Asterisk problems because they accidentally did not have an
indications.conf present and loaded. Now, this code is in the core,
and Asterisk will fail to start without indications configuration.
There was one part of res_indications, the dialplan applications, which did
belong in a module, and have been moved to a new module, app_playtones.
2) Object management has been significantly changed. Tone zones are now
managed using astobj2, and it is no longer possible to crash Asterisk by
issuing a reload that destroys tone zones while they are in use.
3) The API documentation has been filled out.
4) The API has been updated to follow our naming conventions.
5) Various bits of code throughout the tree have been updated to account
for the API update.
6) Configuration parsing has been mostly re-written.
7) "Code cleanup"
The code is from svn/asterisk/team/russell/indications/.
Review: http://reviewboard.digium.com/r/149/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines
Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if
Asterisk runs as a non-root user and the administrator does a 'restart now',
Asterisk loses the ability to set QOS on packets.
(closes issue #14004)
Reported by: nemo
Patches:
20090105__bug14004.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r169722 | tilghman | 2009-01-21 15:02:32 -0600 (Wed, 21 Jan 2009) | 8 lines
Extra NULLs in the output cause some terminal types to abort in the middle of
a color code, causing terminal weirdness.
(closes issue #14130)
Reported by: coolmig
Patches:
20090121__bug14130.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, coolmig
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Spotted and fix provided by ys
(closes issue #14129)
Reported by: ys
Tested by: ys
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@169369 f38db490-d61c-443f-a65b-d21fe96a405b
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The HW_PHYSMEM64 is only available in latest OpenBSD and/or amd64 versions of OpenBSD.
Use HW_PHYSMEM when HW_PHYSMEM64 is not available.
(closes issue #14129)
Reported by: ys
Patches:
2009011600_physmem64.diff.txt uploaded by mvanbaak (license 7)
Tested by: mvanbaak, jtodd
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@169327 f38db490-d61c-443f-a65b-d21fe96a405b
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This started as work to fix the 'core show sysinfo'
CLI command but while working on it oej
pointed out that read_credentials did not compile neither.
So while being there, fix that as well.
Thanks for all the testing oej!
(closes issue #14129)
Reported by: ys
Tested by: oej, mvanbaak
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168609 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #14222)
Reported by: jamesgolovich
Patches:
asterisk-hideconnect.diff.txt uploaded by jamesgolovich (license 176)
Tested by: otherwiseguy
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the next character when a non format specifier is used (no %). Otherwise, the while loop looking for the null byte will never exit.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@167125 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164821 f38db490-d61c-443f-a65b-d21fe96a405b
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sysinfo stuff but do have sysctl.
(closes issue #13433)
Reported by: mvanbaak
Patches:
2008121300_sysinfosysctl.diff.txt uploaded by mvanbaak (license 7)
with two free calls replaced with ast_free based on feedback on reviewboard
Review:
http://reviewboard.digium.com/r/91/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r163761 | tilghman | 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008) | 7 lines
Simple fix for Ctrl-C not immediately exiting Asterisk, but also add a
pointer inside editline to look back to asterisk.c, so others don't spend
as much time as I did looking (in the wrong place) for the appropriate
function.
Reported by: ZX81, via the #asterisk-users channel
Fixed by: me (license 14)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r163383 | tilghman | 2008-12-11 17:35:55 -0600 (Thu, 11 Dec 2008) | 9 lines
When a Ctrl-C or Ctrl-D ends a remote console, on certain shells, the terminal
is messed up. By intercepting those events with a signal handler in the remote
console, we can avoid those issues.
(closes issue #13464)
Reported by: tzafrir
Patches:
20081110__bug13464.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
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(closes issue #13969)
Reported by: jtodd
Patches:
20081205__bug13969.diff.txt uploaded by Corydon76 (license 14)
Tested by: mvanbaak, eliel
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@162687 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@162418 f38db490-d61c-443f-a65b-d21fe96a405b
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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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(closes issue #13990)
Reported by: eliel
Patches:
array_len.diff uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@161218 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@160062 f38db490-d61c-443f-a65b-d21fe96a405b
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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
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Export the XML documentation API:
ast_xmldoc_build_synopsis()
ast_xmldoc_build_syntax()
ast_xmldoc_build_description()
ast_xmldoc_build_seealso()
ast_xmldoc_build_arguments()
ast_xmldoc_printable()
ast_xmldoc_load_documentation()
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(Closes issue #13571)
Reported by: tzafrir
Patch '0001-no-need-for-history-on-asterisk-rx.patch' uploaded by tzafrir.
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branch, and add the ones needed for all the new code here too
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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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background, by using the startup flag '-B'.
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@146359 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@142992 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #13306)
Reported by: Corydon76
Patches:
20080814__bug13306__3.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, pkempgen
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files from main/
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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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dedicated
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: http://svn.digium.com/svn/asterisk/trunk@131445 f38db490-d61c-443f-a65b-d21fe96a405b
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opened when running startup commands from cli.conf
(closes issue #13066)
Reported by: eliel
Patches:
asterisk.c.patch uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130854 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #13054)
Reported by: pabelanger
Patches:
ARRAY_LEN.patch2 uploaded by pabelanger (license 224)
Tested by: seanbright
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(closes issue #13001)
Reported by: eliel
Patches:
20080708__bug13001.diff.txt uploaded by Corydon76 (license 14)
20080708__bug13001.diff.txt.fixed uploaded by eliel (license 64)
Tested by: Corydon76, eliel
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129114 f38db490-d61c-443f-a65b-d21fe96a405b
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r123869 | tilghman | 2008-06-19 11:07:23 -0500 (Thu, 19 Jun 2008) | 6 lines
The RDTSC instruction was introduced on the Pentium line of microprocessors,
and is not compatible with certain 586 clones, like Cyrix. Hence, asking for
i386 compatibility was always incorrect. See http://en.wikipedia.org/wiki/RDTSC
(Closes issue #12886)
Reported by tecnoxarxa
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50 ticks per second, and then counts to see how many ticks it actually
gets in a second.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122926 f38db490-d61c-443f-a65b-d21fe96a405b
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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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