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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/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
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(Also fix the typos in the comments)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273641 f38db490-d61c-443f-a65b-d21fe96a405b
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"initialize"
(closes issue #15571)
Reported by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209098 f38db490-d61c-443f-a65b-d21fe96a405b
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Also, the code in this module is horrendous and we should remove it from the
tree. I'm not sure who is supposed to be maintaning this thing, but they
clearly are not. I don't see the sense of leaving it in the main tree. If it
lives *anywhere* it should be in addons.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204143 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15269)
Reported by: contactmayankjain
Patches:
patch.txt uploaded by contactmayankjain (license 740)
memory_leak_stuff.trunk.diff uploaded by dvossel (license 671)
Tested by: contactmayankjain, dvossel
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201678 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #14833)
Reported by: contactmayankjain
Patches:
all_changes.patch uploaded by contactmayankjain (license 740)
slightly modified by me
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197616 f38db490-d61c-443f-a65b-d21fe96a405b
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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/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196072 f38db490-d61c-443f-a65b-d21fe96a405b
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which tend to break my dev-mode build. Not a problem in 1.6.x.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@178870 f38db490-d61c-443f-a65b-d21fe96a405b
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from app_rpt.c.
(closes issue #14435)
Reported by: D_McNaul
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5 lines
This patch solves some compiler complaints
in both 32 and 64-bit environments.
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172441 f38db490-d61c-443f-a65b-d21fe96a405b
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r159025 | tilghman | 2008-11-24 22:50:00 -0600 (Mon, 24 Nov 2008) | 3 lines
System call ioperm is non-portable, so check for its existence in autoconf.
(Closes issue #13863)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@159050 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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app_voicemail and app_queue.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@137055 f38db490-d61c-443f-a65b-d21fe96a405b
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features that will always be present in DAHDI
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134260 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #13082)
Reported by: eliel
Patches:
app_rpt.c.patch uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@131529 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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@125138 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124596 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r124450 | tilghman | 2008-06-20 18:12:33 -0500 (Fri, 20 Jun 2008) | 6 lines
usleep with a value over 1,000,000 is nonportable. Changing to use sleep()
instead. (closes issue #12814)
Reported by: pputman
Patches:
app_rtp_sleep.patch uploaded by pputman (license 81)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124451 f38db490-d61c-443f-a65b-d21fe96a405b
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Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122234 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #10540)
Reported by: spendergrass
Patches:
20080417__bug10540.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r107464 | kpfleming | 2008-03-11 09:53:03 -0500 (Tue, 11 Mar 2008) | 2 lines
fix various other problems found by gcc 4.3
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107466 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r106552 | tilghman | 2008-03-07 00:36:33 -0600 (Fri, 07 Mar 2008) | 6 lines
Safely use the strncat() function.
(closes issue #11958)
Reported by: norman
Patches:
20080209__bug11958.diff.txt uploaded by Corydon76 (license 14)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106553 f38db490-d61c-443f-a65b-d21fe96a405b
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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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AST_STANDARD_APP_ARGS(foo, bar)
(closes issue #11668, reported and patched by mvanbaak)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95994 f38db490-d61c-443f-a65b-d21fe96a405b
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functions in a few documentation places.
(closes issue #11533)
Reported by: IgorG
Patches:
oldmacroclean.v1.diff uploaded by IgorG (license 20)
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were included almost everywhere.
Remove some of the instances.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89424 f38db490-d61c-443f-a65b-d21fe96a405b
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Reproduce with
grep -r #include . | grep -v .svn | grep -v Binary | sort | uniq -c | sort -nr
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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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in the pbx, a 25-30% speed boost. The two updates used, are, first, to merge the ast_exists_extension() and the ast_spawn_extension() where they are called sequentially in a loop in the code, into a slightly upgraded version of ast_spawn_extension(), with a few extra args; and, second, I modified the substitute_variables_helper_full, so it zeroes out the byte after the evaluated string instead of demanding you pre-zero the buffer; I also went thru the code and removed the code that zeroed this buffer before every call to the substitute_variables_helper_full. The first fix provides about a 9% speedup, and the second the rest. These figures come from the 'PIPS' benchmark I describe in blogs, conf. reports, etc.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88166 f38db490-d61c-443f-a65b-d21fe96a405b
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didn't make much sense
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86820 f38db490-d61c-443f-a65b-d21fe96a405b
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Closes issue #11039, as suggested by seanbright.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86536 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@79747 f38db490-d61c-443f-a65b-d21fe96a405b
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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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microsecond, instead of only to the second
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75706 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75183 f38db490-d61c-443f-a65b-d21fe96a405b
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(issue #9984, patches from eliel and dimas)
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