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2010-07-14ast_callerid restructuringrmudgett1-22/+29
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
2010-07-02Fix various typos reported by Lintiantzafrir1-11/+11
(Also fix the typos in the comments) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273641 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24Make app_rpt.c able to compile again.rmudgett1-35/+35
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265367 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10AST-2009-005tilghman1-10/+22
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211539 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-27Fixing typos. Replaces "recieved" with "received" and "initilize" with ↵dbrooks1-1/+1
"initialize" (closes issue #15571) Reported by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209098 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-29Get app_rpt compiling again. I doubt seriously that it actually works.seanbright1-62/+72
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
2009-06-18fixes some memory leaks and redundant conditionsdvossel1-17/+7
(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
2009-05-28Eliminate several needless checks and fix a few memory leakstilghman1-13/+10
(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
2009-05-21Const-ify the world (or at least a good part of it)kpfleming1-2/+2
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
2009-02-26These small fixes prevent compiler warnings with ubuntu 8.10's gcc-4.3.2, ↵murf1-1/+1
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
2009-02-10This patch removes the use of AST_PBX_KEEPALIVEmurf1-1/+1
from app_rpt.c. (closes issue #14435) Reported by: D_McNaul git-svn-id: http://svn.digium.com/svn/asterisk/trunk@174435 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-10More intptr_t work.murf1-6/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@174432 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-10Merged revisions 174369 via svnmerge from murf1-4/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@174370 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-29Merged revisions 172438 via svnmerge from tilghman1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@172441 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-05Janitor, use ARRAY_LEN() when possible.eliel1-4/+2
(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
2008-11-25Merged revisions 159025 via svnmerge from tilghman1-1/+9
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@159050 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiatingtilghman1-3/+3
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
2008-08-10More RSW merges. Everything from apps/ except for the big offendersseanbright1-64/+61
app_voicemail and app_queue. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@137055 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-29build against the now-typedef-free dahdi/user.h, and remove some #ifdefs for ↵kpfleming1-6/+6
features that will always be present in DAHDI git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134260 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-16Janitor project: convert free to ast_freebbryant1-2/+2
(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
2008-06-25Merged revisions 125132 via svnmerge from kpfleming1-6/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@125138 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-22Let app_rpt compile.seanbright1-2/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124596 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-20Merged revisions 124450 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124451 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-12Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. ↵jpeeler1-201/+201
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
2008-05-22A couple more places the frame data change was missed.seanbright1-58/+58
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117950 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-16Bring all app_rpt and chan_usbradio stuff up to datejdixon1-3624/+11353
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116731 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-01Modify TIMEOUT() to be accurate down to the millisecond.tilghman1-7/+9
(closes issue #10540) Reported by: spendergrass Patches: 20080417__bug10540.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115076 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-25Whitespace changes onlytilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114667 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-11Merged revisions 107464 via svnmerge from kpfleming1-6/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107464 | kpfleming | 2008-03-11 09:53:03 -0500 (Tue, 11 Mar 2008) | 2 lines fix various other problems found by gcc 4.3 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107466 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07Merged revisions 106552 via svnmerge from tilghman1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106553 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-08Merge changes from team/mvanbaak/cli-command-auditrussell1-1/+1
(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! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103171 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-25correct a real problem and silence an annoying compiler warningkpfleming1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@100361 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02Change instances of AST_NONSTANDARD_APP_ARGS(foo, bar, ',') to ↵mmichelson1-8/+8
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
2007-12-13Move usage of the old LOCAL_USER_* macros to the new ast_module_user_* ↵file1-4/+0
functions in a few documentation places. (closes issue #11533) Reported by: IgorG Patches: oldmacroclean.v1.diff uploaded by IgorG (license 20) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92811 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21remove another set of redundant #include "asterisk/options.h"rizzo1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89512 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19include "logger.h" and errno.h from asterisk.h - usage shows that theyrizzo1-2/+0
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
2007-11-16remove a bunch of duplicate includesrizzo1-1/+0
Reproduce with grep -r #include . | grep -v .svn | grep -v Binary | sort | uniq -c | sort -nr git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89348 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16Start untangling header inclusion in a way that does not affectrizzo1-5/+0
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.). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-01This commits the performance mods that give the priority processing engine ↵murf1-1/+1
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
2007-10-22Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former ↵qwell1-6/+6
didn't make much sense git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86820 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-19Convert NEW_CLI to AST_CLI.qwell1-6/+6
Closes issue #11039, as suggested by seanbright. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86536 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :)russell1-121/+148
(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) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-16Don't reload a configuration file if nothing has changed.tilghman1-2/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@79747 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-31Mostly cleanup of documentation to substitute the pipe with the comma, but a ↵tilghman1-2/+2
few other formatting cleanups, too. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77808 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-26Do a massive conversion for using the ast_verb() macrorussell1-26/+13
(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"); git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77299 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-19Fix trunk where I broke it earlier (for ast_strftime branch)tilghman1-5/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75841 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-18Merge in ast_strftime branch, which changes timestamps to be accurate to the ↵tilghman1-4/+5
microsecond, instead of only to the second git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75706 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16Applications no longer need to call ast_module_user_add and ↵file1-3/+0
ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75200 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16It is no longer required for each module that deals with a channel to call ↵file1-1/+0
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
2007-06-24Conversions to ast_debug()russell1-2/+2
(issue #9984, patches from eliel and dimas) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@71338 f38db490-d61c-443f-a65b-d21fe96a405b
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