asterisk

Commit Graph

Author	SHA1	Message	Date
David M. Lee	516dbe86a0	Remove dispatch object allocation from Stasis publishing While looking for areas for performance improvement, I realized that an unused feature in Stasis was negatively impacting performance. When a message is sent to a subscriber, a dispatch object is allocated for the dispatch, containing the topic the message was published to, the subscriber the message is being sent to, and the message itself. The topic is actually unused by any subscriber in Asterisk today. And the subscriber is associated with the taskprocessor the message is being dispatched to. First, this patch removes the unused topic parameter from Stasis subscription callbacks. Second, this patch introduces the concept of taskprocessor local data, data that may be set on a taskprocessor and provided along with the data pointer when a task is pushed using the ast_taskprocessor_push_local() call. This allows the task to have both data specific to that taskprocessor, in addition to data specific to that invocation. With those two changes, the dispatch object can be removed completely, and the message is simply refcounted and sent directly to the taskprocessor. Review: https://reviewboard.asterisk.org/r/2884/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400181 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	9d21631aee	Optimize how Stasis forwards are dispatched This patch optimizes how forwards are dispatched in Stasis. Originally, forwards were dispatched as subscriptions that are invoked on the publishing thread. This did not account for the vast number of forwards we would end up having in the system, and the amount of work it would take to walk though the forward subscriptions. This patch modifies Stasis so that rather than walking the tree of forwards on every dispatch, when forwards and subscriptions are changed, the subscriber list for every topic in the tree is changed. This has a couple of benefits. First, this reduces the workload of dispatching messages. It also reduces contention when dispatching to different topics that happen to forward to the same aggregation topic (as happens with all of the channel, bridge and endpoint topics). Since forwards are no longer subscriptions, the bulk of this patch is simply changing stasis_subscription objects to stasis_forward objects (which, admittedly, I should have done in the first place.) Since this required me to yet again put in a growing array, I finally abstracted that out into a set of ast_vector macros in asterisk/vector.h. Review: https://reviewboard.asterisk.org/r/2883/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	e88afe2022	Taskprocessor optimization; switch Stasis to use taskprocessors This patch optimizes taskprocessor to use a semaphore for signaling, which the OS can do a better job at managing contention and waiting that we can with a mutex and condition. The taskprocessor execution was also slightly optimized to reduce the number of locks taken. The only observable difference in the taskprocessor implementation is that when the final reference to the taskprocessor goes away, it will execute all tasks to completion instead of discarding the unexecuted tasks. For systems where unnamed semaphores are not supported, a really simple semaphore implementation is provided. (Which gives identical performance as the original taskprocessor implementation). The way we ended up implementing Stasis caused the threadpool to be a burden instead of a boost to performance. This was switched to just use taskprocessors directly for subscriptions. Review: https://reviewboard.asterisk.org/r/2881/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400178 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	48d90d947d	Stasis performance improvements This patch addresses several performance problems that were found in the initial performance testing of Asterisk 12. The Stasis dispatch object was allocated as an AO2 object, even though it has a very confined lifecycle. This was replaced with a straight ast_malloc(). The Stasis message router was spending an inordinate amount of time searching hash tables. In this case, most of our routers had 6 or fewer routes in them to begin with. This was replaced with an array that's searched linearly for the route. We more heavily rely on AO2 objects in Asterisk 12, and the memset() in ao2_ref() actually became noticeable on the profile. This was #ifdef'ed to only run when AO2_DEBUG was enabled. After being misled by an erroneous comment in taskprocessor.c during profiling, the wrong comment was removed. Review: https://reviewboard.asterisk.org/r/2873/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	1f908c58d0	res_pjsip: crash when using localnet and external_signaling_address options There was a collision of mod_data use on the transaction between using a nat hook and an session response callback. During state change it was assumed what was in the mod_data was nothing or the response callback. However, it was possible for it to also contain a nat hook thus resulting in a bad cast and a crash. Added the ability to store multiple data elements in mod_data via a hash table. In this instance, mod_data now stores a hash table of the two values that can be retrieved using an associated string key. (closes issue ASTERISK-22394) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/2843/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	d865190a7e	astobj2: Remove OBJ_CONTINUE support. OBJ_CONTINUE was a strange feature that came into the world under suspicious circumstances to support an abuse of the ao2_container by chan_iax2. Since chan_iax2 no longer uses OBJ_CONTINUE, it is safe to remove it. The simplified code should help performance slightly and make understanding the code easier. Review: https://reviewboard.asterisk.org/r/2887/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399937 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	e087610307	Restore usefulness of the CEL Peer field This change makes the CEL peer field useful again for BRIDGE_ENTER and BRIDGE_EXIT events and fills the field with a comma-separated list of all channels in the bridge other than the channel that is entering or exiting the bridge. Review: https://reviewboard.asterisk.org/r/2840/ (closes issue ASTERISK-22393) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399912 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	a8b4e59dbc	pjsip: race condition in registrar While handling a registration request a race condition could occur if/when two+ clients registered at the same time. This happened when one request obtained a copy of the current contacts for an AOR and another request did the same before the first request updated. Thus the second would update and overwrite the first (or vice-versa depending on which actually updated first). In the case of it being the same contact two "add" events would be raised. pjsip registration handling is now serialized to alleviate this issue. (closes issue AST-1213) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2860/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	c4fdbd079c	Change the "external_media_address" PJSIP endpoint option to "media_address". The endpoint option does not apply to communication with external entities. Rather, the option is applied to all communications with the endpoint. The external_media_address transport configuration option may override the endpoint option if it turns out that we are going to be communicating with an external entity. Two things of note: 1) I have not updated the XML documentation. This is being taken care of by Rusty as part of his work on issue ASTERISK-22405 2) This commit is likely to cause testsuite failures since there are tests that use the external_media_address endpoint option, and they will need to be changed over. Well, I'm planning to get that updated ASAP after this commit. (closes issue ASTERISK-22528) reported by Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399283 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	2e17814c28	Fix doxygen to use correct units of features.conf options. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	edb351ebf6	Switch transferdigittimeout to be configured as seconds instead of milliseconds. This was an unintentional consequence of the update of features.conf to use the config framework in Asterisk 12. Thanks to Marco Signorini on the Asterisk developers list for pointing out the problem. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	74c9781273	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399136 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	66834ff561	Create more accurate Contact headers for dialogs when we are the UAS. (closes issue AST-1207) reported by John Bigelow Review: https://reviewboard.asterisk.org/r/2842 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@399083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	ffd49f6b2d	Fix DEBUG_THREADS when lock is acquired in __constructor__ This patch fixes some long-standing bugs in debug threads that were exacerbated with recent Optional API work in Asterisk 12. With debug threads enabled, on some systems, there's a lock ordering problem between our mutex and glibc's mutex protecting its module list (Ubuntu Lucid, glibc 2.11.1 in this instance). In one thread, the module list will be locked before acquiring our mutex. In another thread, our mutex will be locked before locking the module list (which happens in the depths of calling backtrace()). This patch fixes this issue by moving backtrace() calls outside of critical sections that have the mutex acquired. The bigger change was to reentrancy tracking for ast_cond_{timed,}wait, which wrongly assumed that waiting on the mutex was equivalent to a single unlock (it actually suspends all recursive locks on the mutex). (closes issue ASTERISK-22455) Review: https://reviewboard.asterisk.org/r/2824/ ........ Merged revisions 398648 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398649 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398651 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	994a0da4d9	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	7e84877b2c	Fix graceful shutdown crash. The cleanup code for optional_api needs to happen after all of the optional API users and providers have unused/unprovided. Unfortunately, regsitering the atexit() handler at the beginning of main() isn't soon enough, since module destructors run after that. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398149 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	8cc162df5e	Add a reloadable option for sorcery type objects Some configuration objects currently won't place nice if reloaded. Specifically, in this case the pjsip transport objects. Now when registering an object in sorcery one may specify that the object is allowed to be reloaded or not. If the object is set to not reload then upon reloading of the configuration the objects of that type will not be reloaded. The initially loaded objects of that type however will remain. While the transport objects will not longer be reloaded it is still possible for a user to configure an endpoint to an invalid transport. A couple of log messages were added to help diagnose this problem if it occurs. (closes issue ASTERISK-22382) Reported by: Rusty Newton (closes issue ASTERISK-22384) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/2807/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@398139 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	5474e48966	optional_api: Fix linking problems between modules that export global symbols With the new work in Asterisk 12, there are some uses of the optional_api that are prone to failure. The details are rather involved, and captured on [the wiki][1]. This patch addresses the issue by removing almost all of the magic from the optional API implementation. Instead of relying on weak symbol resolution, a new optional_api.c module was added to Asterisk core. For modules providing an optional API, the pointer to the implementation function is registered with the core. For modules that use an optional API, a pointer to a stub function, along with a optional_ref function pointer are registered with the core. The optional_ref function pointers is set to the implementation function when it's provided, or the stub function when it's now. Since the implementation no longer relies on magic, it is now supported on all platforms. In the spirit of choice, an OPTIONAL_API flag was added, so we can disable the optional_api if needed (maybe it's buggy on some bizarre platform I haven't tested on) The AST_OPTIONAL_API() macros themselves remained unchanged, so existing code could remain unchanged. But to help with debugging the optional_api, the patch limits the #include of optional API's to just the modules using the API. This also reduces resource waste maintaining optional_ref pointers that aren't used. Other changes made as a part of this patch: The stubs for http_websocket that wrap system calls set errno to ENOSYS. * res_http_websocket now properly increments module use count. * In loader.c, the while() wrappers around dlclose() were removed. The while(!dlclose()) is actually an anti-pattern, which can lead to infinite loops if the module you're attempting to unload exports a symbol that was directly linked to. * The special handling of nonoptreq on systems without weak symbol support was removed, since we no longer rely on weak symbols for optional_api. [1]: https://wiki.asterisk.org/wiki/x/wACUAQ (closes issue ASTERISK-22296) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2797/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	eae2405dff	ARI: Implement /recordings/stored API's his patch implements the ARI API's for stored recordings. While the original task only specified deleting a recording, it was simple enough to implement the GET for all recordings, and for an individual recording. The recording playback operation was modified to use the same code for accessing the recording as the REST API, so that they will behave consistently. There were several problems with the api-docs that were also fixed, bringing the ARI spec in line with the implementation. There were some 'wishful thinking' fields on the stored recording model (duration and timestamp) that were removed, because I ended up not implementing a metadata file to go along with the recording to store such information. The GET /recordings/live operation was removed, since it's not really that useful to get a list of all recordings that are currently going on in the system. (At least, if we did that, we'd probably want to also list all of the current playbacks. Which seems weird.) (closes issue ASTERISK-21582) Review: https://reviewboard.asterisk.org/r/2693/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e19173ca53	Match use of ast_free() with ast_calloc() and add some curly braces. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	7a60a318ce	bridging: Fix a livelock with local channel optimization. Use a better means of waking up the bridge channel thread. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397650 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	1888186add	Fix the config_options_test The config options test requires the entire configuration item to be transparent from the documentation system. So we let it do that too. As an aside, please do not use this power for evil. Documentation is your friend, and you really should document your configurations. Hiding your module's configuration information from the system attempting to enforce some sanity in the universe is something only a Bond villain would contemplate. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@397628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	dd33217762	Add the bucket API. Bucket is a URI based API for the creation, retrieval, updating, and deletion of "buckets" and files contained within them. Review: https://reviewboard.asterisk.org/r/2715/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	cd7436a6c0	Fix a bug where the argc value was passed as no_doc when registering custom sorcery types. This also adds a _nodoc equivalent. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397599 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	6ebfac8e70	Handle DTMF and hold wrapup when a channel leaves the bridging system. DTMF start/end and hold/unhold events have state because a DTMF begin event and hold event must be ended by something. The following cases need to be handled when a channel is moved around in the system. * When a channel leaves a bridge it may owe a DTMF end event to the bridge. * When a channel leaves a bridge it may owe an UNHOLD event to the bridge. (This case is explicitly ignored because things like transfers need explicit control over this.) * When a channel leaves the bridging system it may need to simulate a DTMF end event to the channel. * When a channel leaves the bridging system it may need to simulate an UNHOLD event to the channel. The patch also fixes the following: * Fixes playing a file and restarting MOH using the latest MOH class used. (closes issue ASTERISK-22043) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2791/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	46b9e5450f	Fix memory corruption when trying to get "core show locks". Review https://reviewboard.asterisk.org/r/2580/ tried to fix the mismatch in memory pools but had a math error determining the buffer size and didn't address other similar memory pool mismatches. * Effectively reverted the previous patch to go in the same direction as trunk for the returned memory pool of ast_bt_get_symbols(). * Fixed memory leak in ast_bt_get_symbols() when BETTER_BACKTRACES is defined. * Fixed some formatting in ast_bt_get_symbols(). * Fixed sig_pri.c freeing memory allocated by libpri when MALLOC_DEBUG is enabled. * Fixed __dump_backtrace() freeing memory from ast_bt_get_symbols() when MALLOC_DEBUG is enabled. * Moved __dump_backtrace() because of compile issues with the utils directory. (closes issue ASTERISK-22221) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2778/ ........ Merged revisions 397525 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397528 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	7a581537e8	ARI: Correct error codes for bridge operations This patch adds error checking to ARI bridge operations, when adding/removing channels to/from bridges. In general, the error codes fall out as follows: * Bridge not found - 404 Not Found * Bridge not in Stasis - 409 Conflict * Channel not found - 400 Bad Request * Channel not in Stasis - 422 Unprocessable Entity * Channel not in this bridge (on remove) - 422 Unprocessable Entity (closes issue ASTERISK-22036) Review: https://reviewboard.asterisk.org/r/2769/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	4d348e853c	Add pass through support for Opus and VP8; Opus format attribute negotiation This patch adds pass through support for Opus and VP8. That includes: * Format attribute negotiation for Opus. Note that unlike some other codecs, the draft RFC specifies having spaces delimiting the attributes in addition to ';', so you have "attra=X; attrb=Y". This broke the attribute parsing in chan_sip, so a small tweak was also included in this patch for that. * A format attribute negotiation module for Opus, res_format_attr_opus * Fast picture update for VP8. Since VP8 uses a different RTCP packet number than FIR, this really is specific to VP8 at this time. Note that the format attribute negotiation in res_pjsip_sdp_rtp was written by mjordan. The rest of this patch was written completely by Lorenzo Miniero. Review: https://reviewboard.asterisk.org/r/2723/ (closes issue ASTERISK-21981) Reported by: Tzafrir Cohen patches: asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero (License 6518) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	e31bd332b8	Update config framework/sorcery with types/options without documentation There are times when a configuration option should not have documentation. 1. Some options are registered with a particular object merely as a warning to users. These options aren't even really 'deprecated' - which has its own separate API call - they are actually provided by a different configuration file. The options are merely registered so that the user gets a warning that a different configuration file provides the item. 2. Some object types - most notably some used by modules that use sorcery - are completely internal and should never be shown to the user. 3. Sorcery itself has several 'hidden' fields that should never be shown to a user. This patch updates the configuration framework and sorcery with additional API calls that allow a module to register types as internal and options as not requiring documentation. This bypasses the XML documentation checking. This patch also re-enables the strict XML documentation checking in trunk, as well as updates some documentation that was missing. Review: https://reviewboard.asterisk.org/r/2785/ (closes issue ASTERISK-22359) Reported by: Matt Jordan (closes issue ASTERISK-22112) Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397524 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	21e22310c7	ARI: Music on Hold/Background Music for bridges Adds ARI functions to be able to turn on/off music on hold in a bridge. It actually functions more as a background music without further actions on the bridge since if the rest of the channels in the bridge aren't explicitly muted, they will still be able to communicate. (closes issue ASTERISK-21974) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2688/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 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	13 years ago
Richard Mudgett	c02c1b6f53	Update MOH start/stop routine doxygen. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397482 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 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	13 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	13 years ago
Mark Michelson	8049bf94f7	Handle default body types for SIP event packages in res_pjsip_pubsub Prior to this change, we would reject SUBSCRIBE requests that had no Accept headers. Now event package handlers that handle the default type for the event package indicate that they do so. Therefore, if we have a handler that can handle the default type, we can allow SUBSCRIBEs for the handler's event package that have no Accept headers. (closes issue ASTERISK-22067) reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/2774 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397441 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	20dcc49d2e	Make CEL behavior conform to the documentation This modifies the behavior of the CEL engine to conform to documented behavior for Asterisk 12 as defined on the wiki https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CEL+Specification The primary changes deal with removal of the peer field from function calls since it is no longer directly relevant to the bridging system and removal of the layer of CDR-like business logic that was providing a partial emulation of Asterisk 11 CEL functionality. With this change, there is no longer a distinction between "bridges" and "conferences" and all participation changes are denoted with bridge enter and bridge exit messages. This updates the CEL unit tests to handle these changes and simplifies some of the macros used in the process. This also fixes a segfault when attempting to ref a configuration that failed to load. Review: https://reviewboard.asterisk.org/r/2788/ (issue ASTERISK-21567) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397431 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	b816fe45b6	* Move ast_bridge_channel_setup_features() into bridge_basic.c. * Made application map hooks be removed on a basic bridge personality change. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	855024107c	Set 14400 as the default max bit rate if T38MaxBitRate is not specified If an endpoint fails to include the T38MaxBitRate attribute during negotiation, Asterisk will negotiate a bit rate of 2400 instead of the ITU recommended bit rate of 14400. This patch fixes this by making AST_T38_RATE_14400 the 'default' value of the enum by assigning it a value of 0, such that if an endpoint fails to include the attribute, the default will be 14400. Note that Walter Doekes included the nice comment in frame.h about why we are purposefully assigning AST_T38_RATE_14400 a value of 0. (closes issue ASTERISK-22275) Reported by: Andreas Steinmetz patches: fax-fix.patch uploaded by anstein (License 6523) ........ Merged revisions 397256 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397257 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 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	13 years ago
Mark Michelson	5caa938be2	Localize and rename ACL configuration. This is more-or-less a reversion of previous ACL behavior so that it is more self-contained. ACL sections are now only parsed if res_pjsip_acl.so is loaded. Moreover, the configuration section is now "type=acl" instead of "type=security". The original reason for having ACLs configured in a "type=security" section was to lump ACLs and other security-related items into the same section. The problem is that ACLs really should be in their own sections and there are no other security-related options implemented anyways. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397193 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	56aea1c030	Allow res_parking to be unloadable This change protects accesses of res_parking such that it can unload safely once transient uses of its registered functions are complete. The parking API has been restructured such that its consumers do not have access to the vtable exposed by the parking provider, but instead route through stubs to prevent consumers from holding on to function pointers. This adds calls to all the parking unload functions and moves application loading and unloading into functions in parking_applications.c similar to the rest of the parts of res_parking. Review: https://reviewboard.asterisk.org/r/2763/ (closes issue ASTERISK-22142) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396890 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	d7f1f31270	Refactor CEL to avoid using the event system core This removes usage of the event system for CEL backend data distribution and strips unused pieces out of the event system. Review: https://reviewboard.asterisk.org/r/2732/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396888 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 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	13 years ago
Richard Mudgett	35b9c6a660	Fix CLI "bridge kick <bridge> <channel>" to check if the bridge needs dissolving. SIP/foo -- Local;1==Local;2 -- .... -- Local;1==Local;2 -- SIP/bar Kick a ;1 channel and the chain toward SIP/foo goes away. Kick a ;2 channel and the chain toward SIP/bar goes away. This can leave a local channel chain between the kicked ;1 and ;2 channels that are orphaned until you manually request one of those channels to hangup or request the bridge to dissolve. * Added ast_bridge_kick() as a companion to ast_bridge_remove(). The functional difference is that ast_bridge_kick() may dissolve the bridge as a result of the channel leaving the bridge. * Made CLI "bridge kick <bridge> <channel>" use ast_bridge_kick() instead of ast_bridge_remove() so the bridge can dissolve if needed. * Renamed bridge_channel_handle_hangup() to ast_bridge_channel_kick() and made it accessible to other files. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396877 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	6399836f9b	Fix some doxygen bridging file references. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	745df0fb81	utils.h: Minor formatting tweaks. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396849 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	f29d969a79	Stasis: address refcount races; implementation comments Change r395954 reordered some stasis object destruction, which should have been fine. Unfortunately, it caused some hard to reproduce issues related to objects being accessed after they had been destroyed. The patch in r396329 fixed the destruction order problem; this patch addresses the underlying issue. A few other stasis-related fixes were also added. * Add ref-bumps around areas where objects may get transitively destroyed. (For example, where we lock a topic, unref a subscription, which unrefs the topic, which explodes the topic when we try to unlock it.) * Wrote an extensive doxygen page about Stasis implementation, relationships between objects, lifecycles of objects, how the refcounting works, etc. Many other comments were added, corrected, or cleaned up. * Added an assert to the topic dtor to catch extra ref decrements. * Fixed type used after destruction errors for graceful shutdown in stasis_channels.c. * I added two unit tests in an attempt to catch destruction order issues. Since the underlying cause is a race condition, though, the tests rarely failed even when the code was wrong. * Fixed a leak in stasis_cache_pattern.c. (closes issue ASTERISK-22243) Review: https://reviewboard.asterisk.org/r/2746/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	c43e19e8e5	Prevent heap alloc functions from running out of stack space. When asterisk has run out of memory (for whatever reason), the alloc function logs a message. Logging requires memory. A recipe for infinite recursion. Stop the recursion by comparing the function call depth for sane values before attempting another OOM log message. Review: https://reviewboard.asterisk.org/r/2743/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396822 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 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	13 years ago
Richard Mudgett	c3466db29d	Resolve some BUGBUG comments. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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