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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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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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2. Added retrieving operator names from AuthRsp and exporting them.
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2. Added service type ported number query.
3. Formated code.
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2. Added destination port support.
3. Added new protocols.
4. Added QoS.
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2. Fixed compile warning for UUID.
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Move OSP* applications static documentation to the new AstXML form.
(closes issue #15245)
Reported by: eliel
Patches:
app_osplookup_static_conversion.txt uploaded by lmadsen (license 10)
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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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which tend to break my dev-mode build. Not a problem in 1.6.x.
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in build_tools/menuselect-deps, so that (once it has been written) menuselect can use this information to warn the user when a previously met dependency is no longer met
along the way, change tags used in configure script, menuselect-deps and code for various dependencies to be consistently named
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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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app_voicemail and app_queue.
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Closes issue #11666, patch by Laureano.
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were included almost everywhere.
Remove some of the instances.
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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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didn't make much sense
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Closes issue #11039, as suggested by seanbright.
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(closes issue #10724)
Reported by: eliel
Patches:
chan_skinny.c.patch uploaded by eliel (license 64)
chan_oss.c.patch uploaded by eliel (license 64)
chan_mgcp.c.patch2 uploaded by eliel (license 64)
pbx_config.c.patch uploaded by seanbright (license 71)
iax2-provision.c.patch uploaded by eliel (license 64)
chan_gtalk.c.patch uploaded by eliel (license 64)
pbx_ael.c.patch uploaded by seanbright (license 71)
file.c.patch uploaded by seanbright (license 71)
image.c.patch uploaded by seanbright (license 71)
cli.c.patch uploaded by moy (license 222)
astobj2.c.patch uploaded by moy (license 222)
asterisk.c.patch uploaded by moy (license 222)
res_limit.c.patch uploaded by seanbright (license 71)
res_convert.c.patch uploaded by seanbright (license 71)
res_crypto.c.patch uploaded by seanbright (license 71)
app_osplookup.c.patch uploaded by seanbright (license 71)
app_rpt.c.patch uploaded by seanbright (license 71)
app_mixmonitor.c.patch uploaded by seanbright (license 71)
channel.c.patch uploaded by seanbright (license 71)
translate.c.patch uploaded by seanbright (license 71)
udptl.c.patch uploaded by seanbright (license 71)
threadstorage.c.patch uploaded by seanbright (license 71)
db.c.patch uploaded by seanbright (license 71)
cdr.c.patch uploaded by moy (license 222)
pbd_dundi.c.patch uploaded by moy (license 222)
app_osplookup-rev83558.patch uploaded by moy (license 222)
res_clioriginate.c.patch uploaded by moy (license 222)
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the ast_check_hangup() funciton. This function takes scheduled hangups into
account.
(closes issue #10230, patch by Juggie)
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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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(issue #9926, caio1982)
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guidelines changes
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(Committed from Media Plaza in Utrecht, Netherlands - Open Source VoIP conference)
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external libraries and URLs to these. Please help me add these
references.
We might want to create a similar macro "\linuxpackage" to list
the needed Linux packages in popular distributions.
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constify a bunch of usage strings for CLI commands.
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initialization discards qualifiers from pointer target type"
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without destination protocol infomation.
2. Fix the bug that Asterisk generats wrong dial string (no in IAX2/[username[:password]@]peer[:port][/exten[@context]][/options] format) for IAX.
3. Add support for oh323 channel driver.
4. Re-formate the code.
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patch provided in bugnote, with minor changes.
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