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Update and extend the configuration_file group and enable linking to the application. Update title that was left behind many years ago.
(issue ASTERISK-20259)
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Start adding configuration file linking and pages. Add module loading doxygen block.
(issue ASTERISK-20259)
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Review: https://reviewboard.asterisk.org/r/1773/
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Review: https://reviewboard.asterisk.org/r/1770/
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There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.
This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.
The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.
The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).
Review: https://reviewboard.asterisk.org/r/1655/
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https://origsvn.digium.com/svn/asterisk/branches/1.10
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r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines
Merged revisions 328209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines
Introduce <support_level> tags in MODULEINFO.
This change introduces MODULEINFO into many modules in Asterisk in order to show
the community support level for those modules. This is used by changes committed
to menuselect by Russell Bryant recently (r917 in menuselect). More information about
the support level types and what they mean is available on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines
Fix a bunch of compiler warnings generated by gcc 4.6.0.
Most of these are -Wunused-but-set-variable, but there were a few others
mixed in here, as well.
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This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal. For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs. Functionally
no change in behavior should be present in this patch. Thanks to twilson
and russell for all the time they spent reviewing these changes.
Review: https://reviewboard.asterisk.org/r/1083/
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r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul 2010) | 8 lines
Total analysis time error with SIP and silence suppression
When using app_amd with SIP providers that have silence
suppression on, the iTotalTime count increases exponentially.
(closes issue #17656)
Reported by: juls
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Expand the ani field in ast_party_caller and ast_party_connected_line to
an ast_party_id.
This is an extension to the ast_callerid restructuring patch in review:
https://reviewboard.asterisk.org/r/702/
Review: https://reviewboard.asterisk.org/r/744/
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The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.
Eliminate struct ast_callerid and replace it with the following struct
organization:
struct ast_party_name {
char *str;
int char_set;
int presentation;
unsigned char valid;
};
struct ast_party_number {
char *str;
int plan;
int presentation;
unsigned char valid;
};
struct ast_party_subaddress {
char *str;
int type;
unsigned char odd_even_indicator;
unsigned char valid;
};
struct ast_party_id {
struct ast_party_name name;
struct ast_party_number number;
struct ast_party_subaddress subaddress;
char *tag;
};
struct ast_party_dialed {
struct {
char *str;
int plan;
} number;
struct ast_party_subaddress subaddress;
int transit_network_select;
};
struct ast_party_caller {
struct ast_party_id id;
char *ani;
int ani2;
};
The new organization adds some new information as well.
* The party name and number now have their own presentation value that can
be manipulated independently. ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.
* The party name and number now have a valid flag. Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.
* The party name now has a character set value. SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.
* The dialed party now has a numbering plan value that could be useful to
have available.
The various channel drivers will need to be updated to support the new
core features as needed. They have simply been converted to supply
current functionality at this time.
The following items of note were either corrected or enhanced:
* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.
* CALLERPRES() is now deprecated because the name and number have their
own presentation values.
* Fixed app_alarmreceiver.c write_metadata(). The workstring[] could
contain garbage. It also can only contain the caller id number so using
ast_callerid_parse() on it is silly. There was also a typo in the
CALLERNAME if test.
* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string. ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string. Then using
ast_shrink_phone_number() could alter it even more.
* Fixed caller ID name and number memory leak in chan_usbradio.c.
* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.
* Protected access to a caller channel with lock in chan_sip.c.
* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk(). Also made save all caller ID data instead of just the name
and number strings.
* Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge()
function.
* Corrected some weirdness with app_privacy.c's use of caller
presentation.
Review: https://reviewboard.asterisk.org/r/702/
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SWP-1229
ABE-2161
* Ensure chan_local.c:local_call() will not leak cid.cid_dnid when
copying.
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r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5 lines
Fix a bug where if you hung up very quickly after calling AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG.
(closes issue #16239)
Reported by: CGMChris
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Reviewboard: https://reviewboard.asterisk.org/r/416/
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This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:
- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments
In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.
Review: https://reviewboard.asterisk.org/r/251/
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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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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)
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(closes issue #11236)
Reported by: philipps
Patches:
20080218__bug11236.diff.txt uploaded by Corydon76 (license 14)
Tested by: philipps
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r101649 | mmichelson | 2008-01-31 18:06:37 -0600 (Thu, 31 Jan 2008) | 9 lines
From bugtracker: "fix totalAnalysisTime to handle periods of no channel activity"
(closes issue #9256)
Reported by: cmaj
Patches:
amd-dont-wait-too-long-for-frames-take3.diff.txt uploaded by cmaj (license 111)
Tested by: cmaj, skygreg, ZX81, rjain
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The channel name is printed in verbose messages
maximumWordLength option added.
Duration of words that do not meet the minimum word duration will be logged
The duration of pre-greeting silence will be logged
Only consider us in the greeting if we actually detected a valid word duration.
(closes issue #11650, reported and patched by davevg)
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(closes issue #11096)
Patches:
pbx_config.c.patch uploaded by moy (license 222)
pbx_dundi.c.patch uploaded by moy (license 222)
pbx_gtkconsole.c.patch uploaded by moy (license 222)
pbx_loopback.c.patch uploaded by moy (license 222)
pbx_realtime.c.patch uploaded by moy (license 222)
pbx_spool.c.patch uploaded by moy (license 222)
app_adsiprog.c.patch uploaded by moy (license 222)
app_alarmreceiver.c.patch uploaded by moy (license 222)
app_amd.c.patch uploaded by moy (license 222)
app_authenticate.c.patch uploaded by moy (license 222)
app_cdr.c.patch uploaded by moy (license 222)
app_zapateller.c.patch uploaded by moy (license 222)
app_zapbarge.c.patch uploaded by moy (license 222)
app_zapras.c.patch uploaded by moy (license 222)
app_zapscan.c.patch uploaded by moy (license 222)
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Reported by: snuffy
Patch by: snuffy
(Closes issue #11547)
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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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(closes issue #11171, reported and patched by blitzrage)
Many thanks!
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few other formatting cleanups, too.
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(closes issue #10277, patches by mvanbaak)
Basically, this changes ...
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3, "Something\n");
to ...
ast_verb(3, "Something\n");
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ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application.
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ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it.
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(issue #9957, patches from mvanbaak, caio1982, critch, and dimas)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47617 | file | 2006-11-14 11:45:57 -0500 (Tue, 14 Nov 2006) | 2 lines
Use LOG_DEBUG to print out the indication that app_amd is using default settings instead of using LOG_NOTICE. This stops needless logging of this information under normal circumstances. (issue #8361 reported by Seb7)
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- restructured build tree and makefiles to eliminate recursion problems
- support for embedded modules
- support for static builds
- simpler cross-compilation support
- simpler module/loader interface (no exported symbols)
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look there, too.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r34087 | kpfleming | 2006-06-14 09:07:53 -0500 (Wed, 14 Jun 2006) | 2 lines
clarify file headers that mention disclaimer usage
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again :-)
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As it turns out, all of these checks were useless, because alloca will never
return NULL.
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As partly documented in loader.c and include/asterisk/module.h,
modules are now expected to return all of their methods and flags
into a structure 'mod_data', and are normally loaded with RTLD_NOW
| RTLD_LOCAL, so symbols are resolved immediately and conflicts
should be less likely. Only in a small number of cases (res_*,
typically) modules are loaded RTLD_GLOBAL, so they can export
symbols.
The core of the change is only the two files loader.c and
include/asterisk/module.h, all the rest is simply adaptation of the
existing modules to the new API, a rather mechanical (but believe
me, time and finger-consuming!) process whose detail you can figure
out by svn diff'ing any single module.
Expect some minor compilation issue after this change, please
report it on mantis http://bugs.digium.com/view.php?id=6968
so we collect all the feedback in one place.
I am just sorry that this change missed SVN version number 20000!
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description() and key() return values
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application wouldn't block indefinitely looking for another frame from that channel. Don't try to do frame size analysis on a frame that isn't voice, only report DEBUG and VERBOSE msgs to the logger channels when the DEBUG and VERBOSE settings are high enough to require it, and some other minor cleanups.
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#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
to these files.
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config to other asterisk samples , bug note 6530
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