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2010-07-16Merged revisions 277182 via svnmerge from pabelanger1-3/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277183 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14Expand the caller ANI field to an ast_party_idrmudgett1-1/+1
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/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276393 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14ast_callerid restructuringrmudgett1-2/+4
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
2010-04-03Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number.rmudgett1-1/+2
SWP-1229 ABE-2161 * Ensure chan_local.c:local_call() will not leak cid.cid_dnid when copying. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256104 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-02Merged revisions 232355 via svnmerge from file1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232356 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Expand codec bitfield from 32 bits to 64 bits.tilghman1-1/+1
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227580 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-07Global var cleanup - constification and removing unused vars.russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199479 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
2008-11-05Add more SeeAlso references based on TFOT.eliel1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154647 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-01Merge changes from team/group/appdocsxmlrussell1-38/+81
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 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153365 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiatingtilghman1-1/+5
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
2008-03-05Create a centralized configuration option for silencethresholdtilghman1-0/+2
(closes issue #11236) Reported by: philipps Patches: 20080218__bug11236.diff.txt uploaded by Corydon76 (license 14) Tested by: philipps git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106072 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-09whitespace fixes only.mvanbaak1-34/+34
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103249 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-01Merged revisions 101649 via svnmerge from mmichelson1-8/+40
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@101650 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-28Some changes to app_amd. mmichelson1-16/+43
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) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95167 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-26Use defined return values in load_module in more places.qwell1-8/+12
(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) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94806 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-14Convert ast_verbose to ast_verb.tilghman1-10/+5
Reported by: snuffy Patch by: snuffy (Closes issue #11547) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92913 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
2007-11-16Start untangling header inclusion in a way that does not affectrizzo1-3/+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
2007-11-06"show application <foo>" changes for clarity.mmichelson1-1/+1
(closes issue #11171, reported and patched by blitzrage) Many thanks! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89044 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-16Don't reload a configuration file if nothing has changed.tilghman1-5/+7
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 ↵tilghman1-3/+3
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() macrorussell1-14/+7
(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
2007-07-16Applications no longer need to call ast_module_user_add and ↵file1-4/+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
2007-06-14Add a massive set of changes for converting to use the ast_debug() macro.russell1-4/+4
(issue #9957, patches from mvanbaak, caio1982, critch, and dimas) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@69327 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-24Doxygen additions, correctionsoej1-0/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@56665 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-14Merged revisions 47617 via svnmerge from file1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47618 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-06Make this module fit the guidelines betterfile1-115/+103
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@42109 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-21merge new_loader_completion branch, including (at least):kpfleming1-19/+12
- 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) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40722 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-20Bug 7399 - Sample config showed [general] as the context, so the app should ↵tilghman1-1/+1
look there, too. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@35185 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-14Merged revisions 34087 via svnmerge from kpfleming1-2/+3
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r34087 | kpfleming | 2006-06-14 09:07:53 -0500 (Wed, 14 Jun 2006) | 2 lines clarify file headers that mention disclaimer usage ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@34090 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-07simplify autoconfig include mechanism (make tholo happy he can use lint ↵kpfleming1-3/+3
again :-) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32846 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-10remove almost all of the checks of the result from ast_strdupa() or alloca().russell1-6/+1
As it turns out, all of these checks were useless, because alloca will never return NULL. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26451 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-14This rather large commit changes the way modules are loaded. rizzo1-15/+7
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! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@20003 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-11normalize code preparing for loader changesrizzo1-3/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@19220 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-08since the module API is changing, it's a good time to const-ify the ↵kpfleming1-2/+2
description() and key() return values git-svn-id: http://svn.digium.com/svn/asterisk/trunk@18552 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-06 More code optimizations. Thanks kpfleming! bweschke1-13/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@18025 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-06 Fix a problem where if the channel was hungup during detection, the ↵bweschke1-29/+54
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. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@18023 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-24Add missingrizzo1-0/+4
#include "asterisk.h" ASTERISK_FILE_VERSION(__FILE__, "$Revision$") to these files. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@14714 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-20Changing syntax once again slightly and standardizingmogorman1-2/+2
config to other asterisk samples , bug note 6530 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10559 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-15remove the uses of the deprecated STANDARD_LOCAL_USERrussell1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10241 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-14Now Kevin will be really happy!mogorman1-17/+17
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10087 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-14changed naming scheme for variables so it matchesmogorman1-20/+22
asterisk standard, changed it so it can take frames of sizes other than 20ms, allowed for the app to reload properly, and finalyl changed sample to general section as to follow standards. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10086 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-14set properties for new files (i need to get this documented)kpfleming1-412/+412
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9960 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-14Commiting 5959 with minor formatting and typomogorman1-0/+412
fixes. Thanks to all those involved. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9918 f38db490-d61c-443f-a65b-d21fe96a405b
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