70 Commits (master)

Author	SHA1	Message	Date
Sean Bright	82bef2ed90	docs: Indent <since> tags. ... Also updates the 'since' of applications/functions that existed before XML documentation was introduced (1.6.2.0).	3 months ago
George Joseph	198b01c536	docs: Add version information to application and function XML elements ... * Do a git blame on the embedded XML application or function element. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The applications and functions defined in the following files were done by hand because the XML was extracted from the C source file relatively recently. * channels/pjsip/dialplan_functions_doc.xml * main/logger_doc.xml * main/manager_doc.xml * res/res_geolocation/geoloc_doc.xml * res/res_stir_shaken/stir_shaken_doc.xml	3 months ago
Naveen Albert	c1b69d460d	callerid.c: Parse previously ignored Caller ID parameters. ... Commit f2f397c1a8 previously made it possible to send Caller ID parameters to FXS stations which, prior to that, could not be sent. This change is complementary in that we now handle receiving all these parameters on FXO lines and provide these up to the dialplan, via chan_dahdi. In particular: * If a redirecting reason is provided, the channel's redirecting reason is set. No redirecting number is set, since there is no parameter for this in the Caller ID protocol, but the reason can be checked to determine if and why a call was forwarded. * If the Call Qualifier parameter is received, the Call Qualifier variable is set. * Some comments have been added to explain why some of the code is the way it is, to assist other people looking at it. With this change, Asterisk's Caller ID implementation is now reasonably complete for both FXS and FXO operation. Resolves: #681	1 year ago
Naveen Albert	bb33925d75	func_callerid: Emit warning if invalid redirecting reason set. ... Emit a warning if REDIRECTING(reason) is set to an invalid reason, consistent with what happens when REDIRECTING(orig-reason) is set to an invalid reason. Resolves: #683	1 year ago
Naveen Albert	273ad73d99	sig_analog: Add fuller Caller ID support. ... A previous change, ASTERISK_29991, made it possible to send additional Caller ID parameters that were not previously supported. This change adds support for analog DAHDI channels to now be able to receive these parameters for on-hook Caller ID, in order to enhance the usability of CPE that support these parameters. Resolves: #94 ASTERISK-30331 UserNote: Additional Caller ID properties are now supported on incoming calls to FXS stations, namely the redirecting reason and call qualifier.	2 years ago
Naveen Albert	9b50bec598	func_callerid: Warn about invalid redirecting reason. ... Currently, if a user attempts to set a Caller ID related function to an invalid value, a warning is emitted, except for when setting the redirecting reason. We now emit a warning if we were unable to successfully parse the user-provided reason. ASTERISK-30332 #close Change-Id: Ic341f5d5f7303b6f1115549be64db58a85944f5a	2 years ago
Jaco Kroon	41389bfdbd	func_callerid+res_agi: Fix compile errors related to -Werror=zero-length-bounds ... Change-Id: I75152cece8a00b7523d542e5ac22796f9595692b Signed-off-by: Jaco Kroon <jaco@uls.co.za>	4 years ago
Sean Bright	056ca07449	func_callerid: Remove deprecated CALLERPRES() function. ... Change-Id: Ia1b2b386505b3102136dab02c45eaaf09f0f89c5	7 years ago
Corey Farrell	a790ced2e8	func_callerid: Initialize app argument structures. ... This module uses AST_DEFINE_APP_ARGS_TYPE to define struct's instead of directly using AST_DECLARE_APP_ARGS. Initialize the variables declared in this way. Change-Id: If97fbdd8d63a204e2efd498a192effc14e90fb31	7 years ago
Corey Farrell	a6e5bae3ef	Remove ASTERISK_REGISTER_FILE. ... ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966	9 years ago
Richard Mudgett	0bdbf0d882	func_callerid.c: Update REDIRECTING reason documentation. ... Change-Id: I6e8d39b0711110a4bceafa652e58b30465e28386	9 years ago
Walter Doekes	7dd8f89a50	func_callerid: Document that CALLERID(pres) is available. ... CALLERPRES() says that it's deprecated in favor of CALLERID(num-pres) and CALLERID(name-pres). But for channel driver that don't make a distinction between the two (e.g. SIP), it makes more sense to get/set both at once. This change reveals the availability of CALLERID(pres), CONNECTEDLINE(pres), REDIRECTING(orig-pres), REDIRECTING(to-pres) and REDIRECTING(from-pres). ASTERISK-25373 #close Change-Id: I5614ae4ab7d3bbe9c791c1adf147e10de8698d7a	10 years ago
Matt Jordan	4a58261694	git migration: Refactor the ASTERISK_FILE_VERSION macro ... Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e	10 years ago
Corey Farrell	fbe0dfaf44	Fix dialplan function NULL channel safety issues ... (closes issue ASTERISK-23391) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3386/ ........ Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411314 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411315 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	6258bbe7bd	Update Asterisk's CDRs for the new bridging framework ... This patch is the initial push to update Asterisk's CDR engine for the new bridging framework. This patch guts the existing CDR engine and builds the new on top of messages coming across Stasis. As changes in channel state and bridge state are detected, CDRs are built and dispatched accordingly. This fundamentally changes CDRs in a few ways. (1) CDRs are now *very* reflective of the actual state of channels and bridges. This means CDRs track well with what an actual channel is doing - which is useful in transfer scenarios (which were previously difficult to pin down). It does, however, mean that CDRs cannot be 'fooled'. Previous behavior in Asterisk allowed for CDR applications, channels, and other properties to be spoofed in parts of the code - this no longer works. (2) CDRs have defined behavior in multi-party scenarios. This behavior will not be what everyone wants, but it is a defined behavior and as such, it is predictable. (3) The CDR manipulation functions and applications have been overhauled. Major changes have been made to ResetCDR and ForkCDR in particular. Many of the options for these two applications no longer made any sense with the new framework and the (slightly) more immutable nature of CDRs. There are a plethora of other changes. For a full description of CDR behavior, see the CDR specification on the Asterisk wiki. (closes issue ASTERISK-21196) Review: https://reviewboard.asterisk.org/r/2486/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	fdfb3ae5fa	Allow for redirecting reasons to be set to arbitrary strings. ... This allows for the REDIRECTING dialplan function to be used to set the reason to any string. The SIP channel driver has been modified to set the redirecting reason string to the value received in a Diversion header. In addition, SIP 480 response reason text will set the redirecting reason as well. (closes issue AST-942) reported by Malcolm Davenport (closes issue AST-943) reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/2101 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	fb6238899b	Add private representation of caller, connected and redirecting party ids. ... This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	73f48997f9	Add original party id and reason support. ... ISDN ETSI PTP and Q.SIG (And SS7 in future) have support for reporting who was the original redirecting party of a call. * Added support for the original redirecting party and reason to the REDIRECTING function and the system core as well as to the stubbed locations in sig_pri.c. Review: https://reviewboard.asterisk.org/r/1829/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	fc63e07135	Avoid cppcheck warnings; removing unused vars and a bit of cleanup. ... Patch by: junky Review: https://reviewboard.asterisk.org/r/1743/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	a9d607a357	Opaquify ast_channel structs and lists ... Review: https://reviewboard.asterisk.org/r/1773/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	57f42bd74f	ast_channel opaquification of pointers and integral types ... Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Paul Belanger	0e887d1a50	Fixed typo from previous commit ... ........ Merged revisions 341704 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341707 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341713 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	5f5e908b19	Updated documentation for the optional CID parameter with CALLERID ... ........ Merged revisions 341664 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341665 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	a525edea59	Merged revisions 328247 via svnmerge from ... 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	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	18 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