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2010-09-09Merged revisions 285710 via svnmerge from bbryant1-1/+7
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent. (closes issue #16903) Reported by: Nick_Lewis Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@285711 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14ast_callerid restructuringrmudgett1-2/+8
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
2010-07-12Added support for indirect work mode.transnexus1-8/+45
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275551 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-02Fix several XML documentation validate errors.lmadsen1-48/+37
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249892 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-12Updated doc for OSP lookup application.transnexus1-2/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@246382 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-13Updated XML doc for OSP.transnexus1-64/+289
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239624 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-041. Added reporting operator names in AuthReq.transnexus1-26/+107
2. Added retrieving operator names from AuthRsp and exporting them. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@237250 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-291. Updated for OSP Toolkit 3.6.0.transnexus1-592/+685
2. Added service type ported number query. 3. Formated code. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@236756 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-03Replaced two deprecated functions of OSP Toolkit.transnexus1-2/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232771 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-03Added custom info support.transnexus1-1/+35
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232738 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-271. Modified exported variable names.transnexus1-120/+458
2. Added destination port support. 3. Added new protocols. 4. Added QoS. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@231401 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-161. Added SIP Diversion support.transnexus1-283/+323
2. Fixed compile warning for UUID. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@230314 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-13Added full number portability parameter support.transnexus1-34/+93
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@229788 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10AST-2009-005tilghman1-17/+17
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211539 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-07Move OSP* applications static documentation to XML.eliel1-64/+170
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) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199547 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-21Const-ify the world (or at least a good part of it)kpfleming1-5/+5
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
2009-05-01Made security features optional.transnexus1-102/+145
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191418 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-30Added routing number support.transnexus1-1/+11
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191332 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-30Fixed not report source network ID and not export destination network ID issues.transnexus1-1/+11
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191300 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-28Updated for OSP Toolkit 3.5.transnexus1-242/+243
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190830 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-26These small fixes prevent compiler warnings with ubuntu 8.10's gcc-4.3.2, ↵murf1-3/+3
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
2008-11-04improve configure script to remember the previous value of each dependency ↵kpfleming1-1/+1
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 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154151 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-13Everytime a compile fails, a puppy dies.seanbright1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@143034 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
2008-08-10More RSW merges. Everything from apps/ except for the big offendersseanbright1-11/+11
app_voicemail and app_queue. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@137055 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02Update osplookup documentation to use commas instead of pipes.qwell1-4/+4
Closes issue #11666, patch by Laureano. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95888 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-19add missing header filedhubbard1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93827 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
2007-11-16Start untangling header inclusion in a way that does not affectrizzo1-4/+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
2007-10-22Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former ↵qwell1-1/+1
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.qwell1-1/+1
Closes issue #11039, as suggested by seanbright. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86536 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-19Fixed a buffer size issue.transnexus1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86439 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :)russell1-36/+36
(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.tilghman1-5/+11
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@79747 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-01Convert code that checks the _softhangup member of ast_channel directory to userussell1-1/+1
the ast_check_hangup() funciton. This function takes scheduled hangups into account. (closes issue #10230, patch by Juggie) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77858 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16Applications no longer need to call ast_module_user_add and ↵file1-26/+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 ↵file1-2/+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.russell1-219/+119
(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
2007-06-12Completely remove all of the code related to jumping to priority n + 101. yay!russell1-55/+4
(issue #9926, caio1982) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@68970 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-06Issue 9869 - replace malloc and memset with ast_calloc, and other coding ↵tilghman1-12/+12
guidelines changes git-svn-id: http://svn.digium.com/svn/asterisk/trunk@67864 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-31Issue #9842 - Doxygen updates by snuffy. Thanks!oej1-0/+1
(Committed from Media Plaza in Utrecht, Netherlands - Open Source VoIP conference) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66705 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-24Creating new doxygen macro "\extref" to create page that lists oej1-0/+3
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. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@56647 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-06Resolve some pointer signedness compiler warnings in app_osplookup, andrussell1-2/+2
constify a bunch of usage strings for CLI commands. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48307 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-051. Change to remove the compiling warning: "app_osplookup.c:2169: warning: ↵transnexus1-1/+1
initialization discards qualifiers from pointer target type" git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48266 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-151. Fix the bug that Asterisk hangs up the calls if the OSP AuthRsp messages ↵transnexus1-202/+677
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. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47704 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-26fix various spelling mistakes in comments (issue #8237, jmls)russell1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@46339 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-03bug #8076 check option_debug before printing to debug channel.mogorman1-106/+198
patch provided in bugnote, with minor changes. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44253 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-20Warning be gone.file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43371 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-18merge qwell's CLI verbification workkpfleming1-8/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43212 f38db490-d61c-443f-a65b-d21fe96a405b