asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	10d4ed93ff	app_speech_utils: Fix unresolved symbol ast_speech_get_setting(). Fixes regression introduced by -r374096. * Made res_speech.export.in export ast_* symbols instead of specific functions. * Made app_speech_utils.c declare that it is dependent upon res_speech. (issue ASTERISK-17136) Reported by: Richard Kenner ........ Merged revisions 399197 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	2a371cd80b	Restore Dial, Queue, and FollowMe 'I' option support. The Dial, Queue, and FollowMe applications need to inhibit the bridging initial connected line exchange in order to support the 'I' option. * Replaced the pass_reference flag on ast_bridge_join() with a flags parameter to pass other flags defined by enum ast_bridge_join_flags. * Replaced the independent flag on ast_bridge_impart() with a flags parameter to pass other flags defined by enum ast_bridge_impart_flags. * Since the Dial, Queue, and FollowMe applications are now the only callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the calling contract to require the initial COLP exchange to already have been done by the caller. * Made all callers of ast_bridge_impart() check the return value. It is important. As a precaution, I also made the compiler complain now if it is not checked. * Did some cleanup in parking_tests.c as a result of checking the ast_bridge_impart() return value. An independent, but associated change is: * Reduce stack usage in ast_indicate_data() and add a dropping redundant connected line verbose message. (closes issue ASTERISK-22072) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/2845/ ........ Merged revisions 399136 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	0ffcd11380	Fix several crashes in MeetMeAdmin This change ensures that MeetMeAdmin commands requiring a user actually get a user and fixes another issue where an extra dereference could occur for a last-entered user being ejected if a user identifier was also provided. (closes issue ASTERISK-21907) Reported by: Alex Epshteyn Review: https://reviewboard.asterisk.org/r/2844/ ........ Merged revisions 399033 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 399034 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 399035 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	4e3f78ad7b	'queue add member' help text correction You are adding dial strings to the queue, not channels. An aribitrary string could be used, but you are typically referencing a channel. Correcting the command help text. (issue ASTERISK-22263) (closes issue ASTERISK-22263) Reported By: Rusty Newton ........ Merged revisions 398884 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398885 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398886 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	f5ae5e27c8	astobj2: Add warn unused attribute to some functions. * Fixed resulting warnings with improper use of ao2_global_obj_replace(). * Made a couple uses of ao2_global_obj_replace_unref(x, NULL) into the equivalent and more appropriate ao2_global_obj_release() call. ........ Merged revisions 398533 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398557 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	64924d72bf	app_voicemail: Fix leaking config objects when msg_id doesn't match (issues ASTERISK-22414) Reported by: Corey Farrell Patch: test_voicemail_api-leaks-11.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 398281 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398283 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	87bedd0695	Fix memory leaks (closes issue ASTERISK-22368) Reported by: Corey Farrell Patches: issueA22368_mixmonitor_free_filename.patch uploaded by wdoekes (license 5674) ........ Merged revisions 398004 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 398011 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398016 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398017 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	d7b9a702d8	Verbose logging discrepancies Refactored cases where a combination of ast_verbose/options_verbose were present. Also in general tried to eliminate, in as many places as possible, where the options_verbose global variable was being used. Refactored the way local and remote consoles handle verbose message logging in an attempt to solve the various discrepancies that sometimes would show between the two. (closes issue AST-1193) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/2798/ ........ Merged revisions 397948 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 397958 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397959 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	0bc2a77365	Multiple revisions 397921-397922 ........ r397921 \| mmichelson \| 2013-08-29 10:42:10 -0500 (Thu, 29 Aug 2013) \| 6 lines Resolve assumptions that bridge snapshots would be non-NULL for transfer stasis events. Attempting to transfer an unbridged call would result in crashes in either CEL code or in the conversion to AMI messages. ........ r397922 \| mmichelson \| 2013-08-29 10:42:29 -0500 (Thu, 29 Aug 2013) \| 3 lines Remove extra debug message. ........ Merged revisions 397921-397922 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397923 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	c25c093c67	Minor tweaks with ast_moh_start() callers. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	7b032c1adb	Add SayAlphaCase and similar functionality for AGI This adds a new dialplan application, SayAlphaCase, that performs much the same function as SayAlpha except that it takes additional options which allow the user to specify whether the case of each letter should be announced for uppercase, lowercase, or all letters. Similar functionality has been added to the SAY ALPHA AGI command via an optional parameter. Original Patch by: Kevin Scott Adams Reported by: Kevin Scott Adams Review: https://reviewboard.asterisk.org/r/2725/ (closes issue ASTERISK-20782) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	477dea4661	Bridge API: Set a cause code on a channel when it is ejected from a bridge. The cause code needs to be passed from the disconnecting channel to the bridge peers if the disconnecting channel dissolves the bridge. * Made the call to an app_agent_pool agent disconnect with the busy cause code if the agent does not ack the call in time or hangs up before acking the call. (closes issue ASTERISK-22042) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2772/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	8300c9aaaf	Remove set but unused variable 'meid'. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	00baddb906	Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d213dfa30f	Fix several interrelated issues dealing with the holding bridge technology. * Added an option flags parameter to interval hooks. Interval hooks now can specify if the callback will affect the media path or not. * Added an option flags parameter to the bridge action custom callback. The action callback now can specify if the callback will affect the media path or not. * Made the holding bridge technology reexamine the participant idle mode option whenever the entertainment is restarted. * Fixed app_agent_pool waiting agents needlessly starting and stopping MOH every second by specifying the heartbeat interval hook as not affecting the media path. * Fixed app_agent_pool agent alert from restarting the MOH after the alert beep. The agent entertainment is now changed from MOH to silence after the alert beep. * Fixed holding bridge technology to defer starting the entertainment. It was previously a mixture of immediate and deferred. * Fixed holding bridge technology to immediately stop the entertainment. It was previously a mixture of immediate and deferred. If the channel left the bridging system, any deferred stopping was discarded before taking effect. * Miscellaneous holding bridge technology rework coding improvements. Review: https://reviewboard.asterisk.org/r/2761/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	430bb3bfb3	Let Queue wrap up time influence member availability Queue members who happen to be in multiple queues at the same time may not have any wrap up time. This problem occurred due to a code change in Asterisk 11.3.0 that unified device state tracking of Queue members in multiple Queues (which fixed some other problems, but unfortunately caused this one). This patch fixes the behavior by having the is_member_available function check the queue's wrap up time and the time of the member's last call, such that for a particular queue, the member won't be considered available if their last call is within the wrap up time. (closes issue ASTERISK-22189) Reported by: Tony Lewis Tested by: Tony Lewis ........ Merged revisions 396948 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	bfcfa2728f	Resolve conflicts between CONFFLAG_DONT_DENOISE and CONFFLAG_INTROUSER_VMREC When r382230 added an option to not denoise the MeetMe conference (if a user had a channel whose format's sample rate changed frequently, for example), the value added was the maximum allowed value for the constants that define the options for MeetMe in 1.8. Not so in 11 - unfortunately, the option CONFFLAG_DONT_DENOISE conflicts with CONFFLAG_INTROUESR_VMREC. This patch fixes that, and also tweaks one of the way in which the constants was declared for consistency. Thanks to Tony Mountifield for pointing out the problem and solution. (closes issue ASTERISK-22269) Reported by: Tony Mountifield ........ Merged revisions 396944 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	59753b1ea1	Strip down the old event system This removes unused code, event types, IE pltypes, and event IE types where possible and makes several functions private that were once public. This includes a renumbering of the remaining event and IE types which breaks binary compatibility with previous versions. The last remaining consumers of the old event system (or parts thereof) are main/security_events.c, res/res_security_log.c, tests/test_cel.c, tests/test_event.c, main/cel.c, and the CEL backends. Review: https://reviewboard.asterisk.org/r/2703/ (closes issue ASTERISK-22139) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	e47d3db365	Doxygen comment tweaks. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396857 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	e35860f954	Changed some BUGBUG tags to associated JIRA issue tags. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396792 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	82ba10bb47	Fix feature_attended_transfer test The feature_attended_transfer test is failing due to Asterisk not passing DTMF in the bridges created for internal attended transfers. This sets the features initialization routine to set this flag by default and adjusts the basic bridge and confbridge's use of the bridging system accordingly as per Richard's suggestion instead of adjusting this individual case. This change allows the necessary DTMF to pass through the attended transfer bridge and complete the test successfully. Review: https://reviewboard.asterisk.org/r/2759/ (closes issue ASTERISK-22222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	08991ac281	app_bridgewait: Inhibit local channel optimizations to the bridge. Holding bridges can allow local channel move/swap optimization to the bridge. However, we cannot allow it for the BridgeWait holding bridge because the call will lose the channel roles and dialplan location as a result. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396695 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	6eec8a44e7	Update documentation for ConfBridge with some additional markup Add some additional markup for items that needed it, e.g., replaceable tags, literal tags, etc. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3f724fa493	Make bridge snapshots use prefixes. * Changed ast_manager_build_bridge_state_string() to assume an empty prefix string just like ast_manager_build_channel_state_string(). * Created ast_manager_build_bridge_state_string_prefix() to work just like ast_manager_build_channel_state_string_prefix(). * Made BridgeMerge AMI event use To/From prefixes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	33e7b76d1d	Hide the Surrogate channels from external consumers; kill Masquerade events This patch does three things: 1. It provides a Surrogate channel technology with a consolidated "implementation detail flag" on the channel technology. This tells consumers of Stasis that the creation of this channel is an implementation detail in Asterisk and can be ignored (if they so choose). This consolidates the conference recorder/announcer flags as well - these flags had no additional meaning beyond "ignore this channel please". 2. It modifies allocation of a channel in two ways: (a) If a channel technology can be determined from the name, we set it directly in the allocation routine. This prevents the initial publication of the message from going out with a NULL channel technology where possible. This lets Stasis consumers get the right channel technology on the first publication. (b) It reorganizes allocation to make use of the 'finalized' property on the channel. This was already used to know that a channel had completely finished its construction in the masquerade routine; now we also use it to know whether or not the setting of certain channel properties is occurring during or post construction. The various set routines were modified accordingly as well. 3. The masquerade event is now dead, Jim. It no longer served any purpose whatsoever - if you perform a call pickup you'll get a Pickup event; if you perform an attended transfer you will still get those events; if you steal a channel to put it elsewhere you'll get the corresponding NewExten or BridgeEnter events. Review: https://reviewboard.asterisk.org/r/2740 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396392 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	200ed6a405	Perform Ring-No-Answer checks before processing Hangup logic The rna() routine will raise a Stasis message involving both the caller and the agent. This doesn't work so well if we already hung up the agent channel, as the channel doesn't quite exist. Not surprisingly, this will crash. This patch properly runs the rna subroutine (performing all of the Ring-No-Answer logic) prior to hanging up the agent channel. (closes issue ASTERISK-22258) Reported by: Kiril Valchev Tested by: Kiril Valchev git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	860ab29dab	Fixed app_meetme for cache split changes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	c790848794	ARI: Add recording controls This patch implements the controls from ARI recordings. The controls are: * DELETE /recordings/live/{recordingName} - stop recording and discard it * POST /recordings/live/{recordingName}/stop - stop recording * POST /recordings/live/{recordingName}/pause - pause recording * POST /recordings/live/{recordingName}/unpause - resume recording * POST /recordings/live/{recordingName}/mute - mute recording (record silence to the file) * POST /recordings/live/{recordingName}/unmute - unmute recording. Since this underlying functionality did not already exist, is was added to app.c by a set of control frames, similar to how playback control works. The pause/mute control frames are toggles, even though the ARI controls are idempotent, to be consistent with the playback control frames. (closes issue ASTERISK-22181) Review: https://reviewboard.asterisk.org/r/2697/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	ccdfe67bf2	Check result of ast_var_assign() calls for memory allocation failure. We try to keep the system running even when all available memory is spent. Review: https://reviewboard.asterisk.org/r/2734/ ........ Merged revisions 396279 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 396287 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	f8622e7c5c	Get rid of ast_bridged_channel() and the bridged_channel field on ast_channels. This commit is smaller than the initial review placed on review board. This is because a change to allow for channel drivers to access parking functionality externally was committed and invalidated quite a few of the changes initially made. (closes issue ASTERISK-22039) reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2717 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396103 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	38236e54a8	Remove dead code from features.c; refactor pickup code into pickup.c This patch does the following: * It moves the pickup code out of features.c and into pickup.c * It removes the vast majority of dead code out of features.c. In particular, this includes the parking code. (issue ASTERISK-22134) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c8a91b5b01	Add queue member paused hints This patch adds the ability in Queue to raise a hint when a member's paused state changes. The hint uses the form 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name} are the name of the queue and the name of the member to subscribe to, respectively. For example: exten => 8501,hint,Queue:sales_pause_mark. Members will show as In Use when paused. Note that the format of the queue pause hint was changed slightly from what is on the issue to accomodate suggestion on the code review. Review: https://reviewboard.asterisk.org/r/2254 (closes issue ASTERISK-20842) Reported by: Philippe Lindheimer patches: qpause-10-378206.diff uploaded by Philippe Lindheimer (license 5519) qpause-11-378206.diff uploaded by Philippe Lindheimer (license 5519) qpause-trunk-378206.diff uploaded by Philippe Lindheimer (license 5519) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	03090a88ba	Fix documentation replication issues This prevents XML documentation duplication by expanding channel and bridge snapshot tags into channel and bridge snapshot parameter sets with a given prefix or defaulting to no prefix. This also prevents documentation from becoming fractured and out of date by keeping all variations of the documentation in template form such that it only needs to be updated once and keeps maintenance to a minimum. Review: https://reviewboard.asterisk.org/r/2708/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	e1b959ccbb	Split caching out from the stasis_caching_topic. In working with res_stasis, I discovered a significant limitation to the current structure of stasis_caching_topics: you cannot subscribe to cache updates for a single channel/bridge/endpoint/etc. To address this, this patch splits the cache away from the stasis_caching_topic, making it a first class object. The stasis_cache object is shared amongst individual stasis_caching_topics that are created per channel/endpoint/etc. These are still forwarded to global whatever_all_cached topics, so their use from most of the code does not change. In making these changes, I noticed that we frequently used a similar pattern for bridges, endpoints and channels: single_topic ----------------> all_topic ^ \| single_topic_cached ----+----> all_topic_cached \| +----> cache This pattern was extracted as the 'Stasis Caching Pattern', defined in stasis_caching_pattern.h. This avoids a lot of duplicate code between the different domain objects. Since the cache is now disassociated from its upstream caching topics, this also necessitated a change to how the 'guaranteed' flag worked for retrieving from a cache. The code for handling the caching guarantee was extracted into a 'stasis_topic_wait' function, which works for any stasis_topic. (closes issue ASTERISK-22002) Review: https://reviewboard.asterisk.org/r/2672/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395954 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	c017d5e6a3	Remove the unsafe bridge parameter from ast_bridge_hook_callback's. Most hook callbacks did not need the bridge parameter. The pointer value could become invalid if the channel is moved to another bridge while it is executing. * Fixed some issues in feature_attended_transfer() as a result. * Reduce the bridge inhibit count in attended_transfer_properties_shutdown() after it has restored the bridge channel hooks. * Removed basic bridge requirement on feature_blind_transfer(). It does not require the basic bridge like feature_attended_transfer(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395574 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	50aba6be36	Improved feature limits interval hook implementaion. * Fixed feature limits to not use special members of struct ast_bridge_features. * Fixed memory leak in off nominal paths of bridge_builtin_set_limits(). * Fixed off nominal path in ast_bridge_features_limits_construct() freeing unallocated memory if it was not called by bridge_builtin_set_limits(). * Made bridge_builtin_interval_features.so unloadable. * Simplified parking's use of its duration interval hook. * Made BridgeWait S option not depend upon another module being loaded. (closes issue ASTERISK-22107) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2701/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	9a46c1d019	Add name argument to BridgeWait() so multiple holding bridges may be used Changes arguments for BridgeWait from BridgeWait(role, options) to BridgeWait(bridge_name, role, options). Now multiple holding bridges may be created and referenced by this application. (closes issue ASTERISK-21922) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2642/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	eec3150a8a	Remove some unnecessary parentheses. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395477 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	cafc115896	A great big renaming patch This patch renames the bridging* files to bridge. This may seem pedantic and silly, but it fits better in line with current Asterisk naming conventions: channel is not "channeling" * monitor is not "monitoring" etc. A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is the act of using a bridge on a set of channels - and the API that fulfills that role is more than just the action. (closes issue ASTERISK-22130) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9d8a5ceb02	Move after bridge callbacks into their own file One more major refactoring to go. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	07d2694f72	Refactor ast_bridge_features struct. * Reduced the number of hook containers to just dtmf_hooks, interval_hooks, and other_hooks. As a result, several functions dealing with the different hook containers could be combined. * Extended the generic hook struct for DTMF and interval hooks instead of using a variant record. * Merged the special talk detector hook into the other_hooks container. * Replaced ast_bridge_features_set_talk_detector() with ast_bridge_talk_detector_hook(). (issue ASTERISK-22107) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395322 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	1d1650f572	Update bridge_channel refactorings; export bridge_ symbol git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	d91dc6d1a8	Perform the initial renaming of the Bridging API This patch does the following: * It pulls out bridge_channel and puts it into its own translation unit * It adds public and protected headers for bridging_channel. Protected functions are appropriate only for the Bridging API and sub-classes of a bridge. (issue ASTERISK-22130) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395253 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	684c83b29b	Add transfer support to CEL This adds CEL support for blind and attended transfers and call pickup. During the course of adding this functionality I noticed that CONF_ENTER, CONF_EXIT, and BRIDGE_TO_CONF events are particularly useless without a bridge identifier, so I added that as well. This adds tests for blind transfers, several types of attended transfers, and call pickup. The extra field in CEL records now consists of a JSON blob whose fields are defined on a per-event basis. Review: https://reviewboard.asterisk.org/r/2658/ (closes issue ASTERISK-21565) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	5a8f32703c	Filter channels used as internal mechanisms This adds new flags to the channel tech properties that flag it as different types of implementation detail used exclusively to provide a feature. Examples of channels that would have these flags include the announcement and recording channels used by confbridge which are the only two marked as such by this patch. Review: https://reviewboard.asterisk.org/r/2633/ (closes issue ASTERISK-21873) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	81f36bee0f	bridge_holding/app_bridgewait: Add new entertainment options This patch adds more entertainment options to holding bridges and the bridge_wait application. Also, holding bridges will now use music on hold as the default entertainment option instead of none. The parameters for app_bridgewait have changed to (role, options) from the previous (options) and the options themselves have changed as well (entertainment options are now contained in an enumerator, role specification is handled by the role parameter, etc) (closes issue ASTERISK-21923) Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/2679/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394731 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	40ce5e0d18	Change ast_hangup() to return void and be NULL safe. Since ast_hangup() is effectively a channel destructor, it should be a void function. * Make the few silly callers checking the return value no longer do so. Only the CDR and CEL unit tests checked the return value. * Make all callers take advantage of the NULL safe change and remove the NULL check before the call. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	4f29b97020	app_confbridge: Eliminate a reference leak for confbridge announcer channels git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	19d8f8c8e4	Add 'kick all' capability to ConfBridge CLI command This patch adds the ability to kick all users out of a conference from the ConfBridge kick CLI command. It is invoked by passing 'all' as the channel parameter to the CLI command, i.e., "confbridge kick <conf> all". Note that this patch was modified slightly to conform to trunk. (closes issue ASTERISK-21827) Reported by: dorianlogan patches: kickall-patch_v2.diff uploaded by dorianlogan (License 6504) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394531 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d43b17a872	Replace chan_agent with app_agent_pool. The ill conceived chan_agent is no more. It is now replaced by app_agent_pool. Agents login using the AgentLogin() application as before. The AgentLogin() application no longer does any authentication. Authentication is now the responsibility of the dialplan. (Besides, the authentication done by chan_agent did not match what the voice prompts asked for.) Sample extensions.conf [login] ; Sample agent 1001 login ; Set COLP for in between calls so the agent does not see the last caller COLP. exten => 1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>) ; Give the agent DTMF transfer and disconnect features when connected to a caller. same => n,Set(CHANNEL(dtmf-features)=TX) same => n,AgentLogin(1001) same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS}) same => n,Hangup() [caller] ; Sample caller direct connect to agent 1001 exten => 800,1,AgentRequest(1001) same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS}) same => n,Hangup() ; Sample caller going through a Queue to agent 1001 exten => 900,1,Queue(agent_q) same => n,Hangup() Sample queues.conf [agent_q] member => Local/800@caller,,SuperAgent,Agent:1001 Under the hood operation overview: 1) Logged in agents wait for callers in an agents holding bridge. 2) Caller requests an agent using AgentRequest() 3) A basic bridge is created, the agent is notified, and caller joins the basic bridge to wait for the agent. 4) The agent is either automatically connected to the caller or must ack the call to connect. 5) The agent is moved from the agents holding bridge to the basic bridge. 6) The agent and caller talk. 7) The connection is ended by either party. 8) The agent goes back to the agents holding bridge. To avoid some locking issues with the agent holding bridge, I needed to make some changes to the after bridge callback support. The after bridge callback is now a list of requested callbacks with the last to be added the only active callback. The after bridge callback for failed callbacks will always happen in the channel thread when the channel leaves the bridging system or is destroyed. (closes issue ASTERISK-21554) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2657/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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4409 Commits (e77fba4b251dec3e56abc2f1e16ab28ace58d5bf)