summaryrefslogtreecommitdiff
path: root/apps/app_osplookup.c
AgeCommit message (Collapse)Author
2011-11-14Increased max number of destinations.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Updated for OSP Toolkit 4.0.0.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338136 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-14Merged revisions 328247 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ 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 ........ 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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-10Fixed the issue caused by EXTEN including user parameters.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14ast_callerid restructuringRichard Mudgett
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: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-12Added support for indirect work mode.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-02Fix several XML documentation validate errors.Leif Madsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-12Updated doc for OSP lookup application.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-13Updated XML doc for OSP.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-041. Added reporting operator names in AuthReq.TransNexus OSP Development
2. Added retrieving operator names from AuthRsp and exporting them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-291. Updated for OSP Toolkit 3.6.0.TransNexus OSP Development
2. Added service type ported number query. 3. Formated code. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-03Replaced two deprecated functions of OSP Toolkit.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-03Added custom info support.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-271. Modified exported variable names.TransNexus OSP Development
2. Added destination port support. 3. Added new protocols. 4. Added QoS. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-161. Added SIP Diversion support.TransNexus OSP Development
2. Fixed compile warning for UUID. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-13Added full number portability parameter support.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-10AST-2009-005Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-07Move OSP* applications static documentation to XML.Eliel C. Sardanons
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: https://origsvn.digium.com/svn/asterisk/trunk@199547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21Const-ify the world (or at least a good part of it)Kevin P. Fleming
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: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-01Made security features optional.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191418 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-30Added routing number support.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-30Fixed not report source network ID and not export destination network ID issues.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-28Updated for OSP Toolkit 3.5.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-26These small fixes prevent compiler warnings with ubuntu 8.10's gcc-4.3.2, ↵Steve Murphy
which tend to break my dev-mode build. Not a problem in 1.6.x. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-04improve configure script to remember the previous value of each dependency ↵Kevin P. Fleming
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: https://origsvn.digium.com/svn/asterisk/trunk@154151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-13Everytime a compile fails, a puppy dies.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@143034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-12Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiatingTilghman Lesher
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: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-10More RSW merges. Everything from apps/ except for the big offendersSean Bright
app_voicemail and app_queue. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-02Update osplookup documentation to use commas instead of pipes.Jason Parker
Closes issue #11666, patch by Laureano. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-19add missing header fileDwayne M. Hubbard
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21remove another set of redundant #include "asterisk/options.h"Luigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19include "logger.h" and errno.h from asterisk.h - usage shows that theyLuigi Rizzo
were included almost everywhere. Remove some of the instances. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16Start untangling header inclusion in a way that does not affectLuigi Rizzo
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: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-22Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former ↵Jason Parker
didn't make much sense git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-19Convert NEW_CLI to AST_CLI.Jason Parker
Closes issue #11039, as suggested by seanbright. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-19Fixed a buffer size issue.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :)Russell Bryant
(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: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-16Don't reload a configuration file if nothing has changed.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-01Convert code that checks the _softhangup member of ast_channel directory to useRussell Bryant
the ast_check_hangup() funciton. This function takes scheduled hangups into account. (closes issue #10230, patch by Juggie) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-16Applications no longer need to call ast_module_user_add and ↵Joshua Colp
ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-16It is no longer required for each module that deals with a channel to call ↵Joshua Colp
ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-14Add a massive set of changes for converting to use the ast_debug() macro.Russell Bryant
(issue #9957, patches from mvanbaak, caio1982, critch, and dimas) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-12Completely remove all of the code related to jumping to priority n + 101. yay!Russell Bryant
(issue #9926, caio1982) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-06Issue 9869 - replace malloc and memset with ast_calloc, and other coding ↵Tilghman Lesher
guidelines changes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-31Issue #9842 - Doxygen updates by snuffy. Thanks!Olle Johansson
(Committed from Media Plaza in Utrecht, Netherlands - Open Source VoIP conference) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-24Creating new doxygen macro "\extref" to create page that lists Olle Johansson
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: https://origsvn.digium.com/svn/asterisk/trunk@56647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-06Resolve some pointer signedness compiler warnings in app_osplookup, andRussell Bryant
constify a bunch of usage strings for CLI commands. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-051. Change to remove the compiling warning: "app_osplookup.c:2169: warning: ↵TransNexus OSP Development
initialization discards qualifiers from pointer target type" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-151. Fix the bug that Asterisk hangs up the calls if the OSP AuthRsp messages ↵TransNexus OSP Development
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: https://origsvn.digium.com/svn/asterisk/trunk@47704 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-26fix various spelling mistakes in comments (issue #8237, jmls)Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46339 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-05 10:51:01 +0000