49 Commits (10e41e37b5031694e3ce663e89b14912816c859c)

Author	SHA1	Message	Date
Paul Belanger	467d40c0b6	Fixed typo from previous commit ... ........ Merged revisions 341704 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341707 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	986c3e6c64	Updated documentation for the optional CID parameter with CALLERID ... ........ Merged revisions 341664 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	7caa2349af	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/branches/1.10@328247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	cf7bbcc4c6	Expand the caller ANI field to an ast_party_id ... 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: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ec37ffbdaf	ast_callerid restructuring ... 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	15 years ago
Mark Michelson	b5d5cc565f	Enhancements to connected line and redirecting work. ... From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a5a0a5f867	Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number. ... SWP-1229 ABE-2161 * Ensure chan_local.c:local_call() will not leak cid.cid_dnid when copying. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	1174a61612	Add support for calling and called subaddress. Partial support for COLP subaddress. ... The Telecom Specs in NZ suggests that SUB ADDRESS is always on, so doing "desk to desk" between offices each with an asterisk box over the ISDN should then be possible, without a whole load of DDI numbers required. (closes issue #15604) Reported by: alecdavis Patches: asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license 585) Some minor modificatons were made. Tested by: alecdavis, rmudgett Review: https://reviewboard.asterisk.org/r/405/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Michiel van Baak	7eac18b09c	add name argument for the CALLERID dialplan function to the xml documentation. ... Pointed out to me on IRC by snuff-home. Thanks git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220629 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	a866a75900	Merge str_substitution branch. ... This branch adds additional methods to dialplan functions, whereby the result buffers are now dynamic buffers, which can be expanded to the size of any result. No longer are variable substitutions limited to 4095 bytes of data. In addition, the common case of needing buffers much smaller than that will enable substitution to only take up the amount of memory actually needed. The existing variable substitution routines are still available, but users of those API calls should transition to using the dynamic-buffer APIs. Reviewboard: http://reviewboard.digium.com/r/174/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191140 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	64a1895f3c	Jcolp pointed out that num will also match number ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	7ed9924348	* Found a couple more places where num/number needed to be done ... so 1.4 upgraders will not have problems. * Added curly braces and minor tweaks. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	c6ebdafd0e	Merged revisions 160703 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160703 | murf | 2008-12-03 13:41:42 -0700 (Wed, 03 Dec 2008) | 11 lines (closes issue #13597) Reported by: john8675309 Patches: patch.13597 uploaded by murf (license 17) Tested by: murf, john8675309 This patch causes the setcid func to update the CDR clid after setting the channel field. I also notice that in trunk, the num/number of 1.4 is left out; I decided to include the option to use either in trunk, so as not to have 1.4 upgraders not to have problems. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5b168ee34b	Merge changes from team/group/appdocsxml ... 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: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	c2d9b9c009	Independent change from branch issue8824 that is not part of COLP. (-r142574 rmudgett) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	63b165dbb9	Merged revisions 146799 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r146799 | tilghman | 2008-10-06 15:52:04 -0500 (Mon, 06 Oct 2008) | 8 lines Dialplan functions should not actually return 0, unless they have modified the workspace. To signal an error (and no change to the workspace), -1 should be returned instead. (closes issue #13340) Reported by: kryptolus Patches: 20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	53a5f22849	Merged revisions 90145 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90145 | russell | 2007-11-28 18:20:34 -0600 (Wed, 28 Nov 2007) | 5 lines This set of changes is to make some callerID handling thread-safe. The ast_set_callerid() function needed to lock the channel. Also, the handlers for the CALLERID() dialplan function needed to lock the channel when reading or writing callerid values directly on the channel structure. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	7e8835e0d7	remove another set of redundant #include "asterisk/options.h" ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	9335ace850	another bunch of include removals (errno.h and asterisk/logger.h) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	fdb7f7ba3d	Start untangling header inclusion in a way that does not affect ... 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	18 years ago
Matthew Fredrickson	f2f66caae1	See if I can fix this borked ANI2 code I added ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85498 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Matthew Fredrickson	4c5dd299bc	Add ANI2 support to func_callerid ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85496 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	55b1ee298e	Merge the dialplan_aesthetics branch. Most of this patch simply converts applications ... using old methods of parsing arguments to using the standard macros. However, the big change is that the really old way of specifying application and arguments separated by a comma will no longer work (e.g. NoOp,foo|bar). Instead, the way that has been recommended since long before 1.0 will become the only method available (e.g. NoOp(foo,bar). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76703 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	9d05ff8ed5	Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	b230918437	Merged revisions 61681 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61681 | tilghman | 2007-04-18 21:45:05 -0500 (Wed, 18 Apr 2007) | 13 lines Merged revisions 61680 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 Apr 2007) | 5 lines Bug 9557 - Specifying the GetVar AMI action without a Channel parameter can cause Asterisk to crash. The reason this needs to be fixed in the functions instead of in AMI is because Channel can legitimately be NULL, such as when retrieving global variables. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61682 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	c59714fbfc	Add CALLERPRES dialplan function and deprecate SetCallerPres application ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	0ddb38ddee	Doxygen update ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	3c1b9fc508	One const, two const. Let's stick with everything else - one const. Plus older versions of GCC don't support double const either. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	37182c873e	finish const-ifying ast_func_read() ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49741 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	527dbe399e	Simplify the if statements used to check to see if the argument was "num" ... or "number". It is not possible to ever reach the second part of this conditional statement. Thanks to my brother, Brett, for pointing this out. :) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Paul Cadach	500353e095	Extend CALLERID() function for "pres" and "ton" values ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44685 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matthew Fredrickson	33ddb53663	Various updates from PCadach's chan_h323-live branch ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	0a27d8bfe5	merge new_loader_completion branch, including (at least): ... - 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: https://origsvn.digium.com/svn/asterisk/trunk@40722 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	472c1ca282	simplify autoconfig include mechanism (make tholo happy he can use lint again :-) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Luigi Rizzo	652ccf5037	more NULL "" equivalence in CLI fields. ... Mark a place where we don't clear a buffer while we probably should. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@21820 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Luigi Rizzo	e43bc6634d	This rather large commit changes the way modules are loaded. ... 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: https://origsvn.digium.com/svn/asterisk/trunk@20003 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	f10f427d49	since the module API is changing, it's a good time to const-ify the description() and key() return values ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	96bf384e7d	major dialplan functions update ... deprecate LANGUAGE() and MUSICCLASS(), in favor of CHANNEL() git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9674 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Kevin P. Fleming	a38a7eec61	build function modules independently (no more pbx_functions.so) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Tilghman Lesher	5323442db4	Bug 5327 - new function FILTER and optional argument to CALLERID ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7614 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Kevin P. Fleming	2c65582b66	remove extraneous svn:executable properties ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Russell Bryant	987b87662c	make app_queue 1.2 jump compliant (issue #5580) ... add missing includes of stdio.h remove some unused and duplicate headers git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7026 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Russell Bryant	3453e3efa5	Doxygen documentation update from oej (issue #5505) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6847 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Kevin P. Fleming	5da915dcfd	update MANY more files with proper copyright/license info (thanks Ian!) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6596 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Russell Bryant	cdf79750c0	squash a warning ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6196 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Russell Bryant	76b6344760	add ability to set or read both callerid name and num in one call to the function ... This one is for you, Mark :-p git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6188 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Kevin P. Fleming	0db4cee117	enable version string for this module, since it's standalone ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5874 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Kevin P. Fleming	2b8338cb52	more file version tags ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Kevin P. Fleming	fa60672d58	add dialplan functions for Caller ID, language and timeouts (bug #4219, with mods) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5679 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago