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2013-08-17Strip down the old event systemKinsey Moore
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
2013-08-16Fix CLI "bridge kick <bridge> <channel>" to check if the bridge needs ↵Richard Mudgett
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
2013-08-16Fix some doxygen bridging file references.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396867 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-16Doxygen comment tweaks.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-16Fix utilities compilation/linking.Richard Mudgett
The horrid structure of the source in the utils directory strikes again. Moved the _ast_mem_backtrace_buffer[] definition from the logical location in utils.c to hashtab.c so the aelparse and conf2ael utilities can link. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-16utils.h: Minor formatting tweaks.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396849 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-16Stasis: address refcount races; implementation commentsDavid M. Lee
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
2013-08-16Improve sounds indexer CLI commandsKinsey Moore
This reworks the CLI commands used to access sounds information from "sounds show[ soundid]" to "core show sounds" and "core show sound <soundid>". This also reworks the "sounds reload" CLI command to fall under normal module reloading ("module reload sounds"). Also, make trunk build when DEBUG_MALLOC is not enabled. Review: https://reviewboard.asterisk.org/r/2745/ (closes issue ASTERISK-22141) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-16Prevent heap alloc functions from running out of stack space.Walter Doekes
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
2013-08-15Bridge: Don't suspend/unspend the channel for interception routines.Richard Mudgett
By their nature, the connected line and redirecting interception routines are not supposed to affect the channel's media. Therefore, they should not suspend and unsuspend the channel while running. The suspend/unsuspend operations could be expensive depending upon the bridge and channel technology involved. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396814 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-15Minor parking cleanup.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-15Parking: Eliminate local channel name hack to get peer channel.Richard Mudgett
(closes issue ASTERISK-22034) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-15Remove early bridge BUGBUG comments. Remove some unneeded features.c comments.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-15Update features.conf.sample atxferdropcall option.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-15Changed some BUGBUG tags to associated JIRA issue tags.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-15Resolve some BUGBUG comments.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-15Remove leading spaces from the CLI command before parsingKinsey Moore
If you've mistakenly put a space before typing in a command, the leading space will be included as part of the command, and the command parser will not find the corresponding command. This patch rectifies that situation by stripping the leading spaces on commands. Review: https://reviewboard.asterisk.org/r/2709/ Patch-by: Tilghman Lesher ........ Merged revisions 396745 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 396746 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-15Remove some dead code dealing with: AST_BRIDGE_REC_CHANNEL_0, ↵Richard Mudgett
AST_BRIDGE_REC_CHANNEL_1, and AST_BRIDGE_IGNORE_SIGS. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-15Fix Bridge API DTMF hook matching for begin and end DTMF events.Richard Mudgett
The Bridge API DTMF hook matching would not deal with DTMF end events only. It required a DTMF begin event to start matching the DTMF hooks. There are many places in Asterisk where code only generates DTMF end events without the corresponding begin event. One such place is the AMI action Atxfer. * Fixed DTMF hook matching if there is a string of DTMF frames in the read queue. We could potentially miss some of them before. * Fixed AMI Atxfer action documentation. (closes issue ASTERISK-22037) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2752/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-15Fix feature_attended_transfer testKinsey Moore
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
2013-08-15Fix deadlocks in chan_sip in REFER and BYE handlingKinsey Moore
This resolves several deadlocks in chan_sip relating to usage of ast_channel_bridge_peer and improves accessibility of lock debugging function calls. Review: https://reviewboard.asterisk.org/r/2756/ (closes issue ASTERISK-22215) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396723 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-15Prevent automagic things from happening to Stasis application bridgesKinsey Moore
This prevents swap optimization, merges, and transfers involving Stasis application bridges. It wouldn't be nice if the bridge you thought you owned disappeared from under you. Reported-by: Richard Mudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-15Remove unsupported channel technology callbacks.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-14chan_vpb: Effectively remove native support. Left enough bread crumbs to be ↵Richard Mudgett
able to convert later if needed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396712 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-14chan_iax2: Conditionally remove native support for now.Richard Mudgett
(issue ASTERISK-21944) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396710 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-14chan_misdn: Effectively remove native support. Left enough bread crumbs to ↵Richard Mudgett
be able to convert later if needed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396703 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-14app_bridgewait: Inhibit local channel optimizations to the bridge.Richard Mudgett
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
2013-08-14Tweak comment for why usleep is used.Joshua Colp
........ Merged revisions 396656 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 396657 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-14Tweak test_hashtab_thrash test to allow the critical threads to execute.Joshua Colp
Depending on certain conditions it was possible for the hashtab counting thread to starve other threads, preventing them from executing in the expected fashion. This change adds a sleep to allow the others to do what they need to do. While this doesn't thrash the hashtab as much as previously, it at least works. (closes issue ASTERISK-22276) Reported by: Matt Jordan ........ Merged revisions 396619 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 396620 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-13chan_sip: Convert 'just did sched_add waitid...' from warning to debug message.Walter Doekes
Patches: reviewboard-2377.patch uploaded by Paul Belanger Review: https://reviewboard.asterisk.org/r/2377/ ........ Merged revisions 396582 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 396583 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396584 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-13chan_sip: Fix IP-addr in warning when rejecting a contact ACL.Walter Doekes
Patches: reviewboard-2155.patch uploaded by Paul Belanger Review: https://reviewboard.asterisk.org/r/2155/ ........ Merged revisions 396579 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 396580 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-13ARI: allow other operations to happen while bridgedDavid M. Lee
This patch changes ARI bridging to allow other channel operations to happen while the channel is bridged. ARI channel operations are designed to queue up and execute sequentially. This meant, though, that while a channel was bridged, any other channel operations would queue up and execute only after the channel left the bridge. This patch changes ARI bridging so that channel commands can execute while the channel is bridged. For most operations, things simply work as expected. The one thing that ended up being a bit odd is recording. The current recording implementation will fail when one attempts to record a channel that's in a bridge. Note that the bridge itself may be recording; it's recording a specific channel in the bridge that fails. While this is an annoying limitation, channel recording is still very useful for use cases such as voice mail, and bridge recording makes up much of the difference for other use cases. (closes issue ASTERISK-22084) Review: https://reviewboard.asterisk.org/r/2726/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-13Missed a spot in r396559David M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-13Fix build warnings when printf a tv_usec.David M. Lee
The debug logs added in r396528 neglected to account for suseconds_t being an int. See r392076 for more info. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-12Add test suite events for when contacts are added or removed from an AORJohn Bigelow
These are needed by the pjsip inbound registration test suite tests. (issue ASTERISK-21833) (issue ASTERISK-21834) (issue ASTERISK-21835) (issue ASTERISK-21837) Review: https://reviewboard.asterisk.org/r/2700/ Review: https://reviewboard.asterisk.org/r/2739/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-12Fix two race conditions and ref counting issue when joining a bridgeMatthew Jordan
These problems were all caught by a test in the Asterisk Test Suite that originated some Local channels and attempted to move the ;2 half of the Local channel into a bridge using the Bridge AMI action. (1) When originating a channel, the Newchannel event is emitted quickly; however, the ;2 channel will not have a pbx thread assigned to it until after the outbound 'dialing' for the ;1 is complete. Thus, there is a period of time where the outside world "knows" of the channel's existence and can influence it but Asterisk has not yet started the dialplan execution thread. If a Bridge AMI action is taken on the channel, the channel appears to be a Dialed channel with no PBX thread; hence, the channel will be imparted into the Bridge by first 'yanking' the channel. At the same time, a race condition can occur after the yank (but before entering the bridge) when ;1 answers and starts a PBX on the ;2. The end result currently is an assertion failure in the Bridging API, as a channel with a PBX is imparted into the Bridge. There's no way to prevent AMI from attempting to Bridge a channel immediately after creation; likewise, holding the channel lock through the entire Dial operation is unwise (and impossible). Instead of treating the presence of a PBX thread as an error, we simply bail out of the adding the channel to the bridge through ast_bridge_impart. The Bridge action will then fail - but we avoid a situation where the channel is both executing a PBX thread and simultaneously being given a separate thread in the bridging system (which would be a "bad thing"). Since imparting a channel with a PBX *can* occur and is not a programming error, the asserts have been removed. (2) When the first condition occurs, we have to take one of two actions: either hangup the yanked channel as it did not enter the bridge, or deref it because we don't own it. We can determine if we own it or not by testing for the presence of the PBX thread. If we hung it up directly, we'd crash. (3) bridge_find_channel does not increase the reference count of the ast_bridge_channel object. The RAII_VAR usage in ast_bridge_add_channel thus created a ticking time bomb in whatever bridge the channel moved into, as the destructor for the ast_bridge_channel object would be called. Review: https://reviewboard.asterisk.org/r/2741/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-12Unlock outgoing dial lock on off nominal pathMatthew Jordan
If the thread servicing the dial request isn't created successfully, the outgoing dial lock will still be held when the function returns. This patch unlocks the lock on this off nominal path. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-10Pipe test output through test object not stdoutMatthew Jordan
Otherwise, it doesn't show up in the automated test failures git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-10Add some debugging when test_hashtab_thrash failsMatthew Jordan
Disabling DEBUG_THREADS caused this test to fail on the 32-bit build agent. Adding some debugging to see why it thinks the test is timing out. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-10Unlock the dial operation lock on a failed dialMatthew Jordan
If a dial operation fails, the pbx_outgoing_attempt routine will exit without first having unlocked the outgoing dial lock. This would be a "bad thing". git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396521 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-09bridge_native_rtp: Remove some unnecessary NULL checks on c1.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-09Don't leak frames when memory is full in autoservice_run.Walter Doekes
Review: https://reviewboard.asterisk.org/r/2566/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-09pbx: Make originate threads indicate dial status when synchronousJonathan Rose
This makes it so that we can detect failures to originate as with earlier versions of Asterisk, which restores the Asterisk 11 behavior for the originate manager action. This was causing the ACL tests for SIP and IAX2 to fail since those tests expected originate failures when ACLs would cause rejections. Also, this patch fixes crashes in chan_sip when ACLs rejected peers during registration verification. (closes issue ASTERISK-22212) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-09bridge_channel: Support the lonely flag and make ARI use it.Jonathan Rose
The lonely flag is an optional flag for bridge channels that will make them leave a bridge when a channel leaves if only lonely channels are in the bridge at that point. This is useful for things like ending recording and playback channels when they cease to be interacting with other channels in the bridge. (closes issue ASTERISK-22117) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2721/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-09Update documentation for ConfBridge with some additional markupMatthew Jordan
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
2013-08-08Fix stasis/core unit test. Should have had the CR/LF.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-08chan_dahdi: create channels at run-timeTzafrir Cohen
This code adds chan_dahdi the command 'dahdi create channels <range>' (where <range> is a single <n>-<m> or 'new') and updates 'dahdi destroy channel' with a similar 'dahdi destroy channels'. It allows DAHDI channels and spans to be added after the initial channel load (without destroying all other channels as in 'dahdi restart'). It also includes some fixes to the D-Channel / span destruction code (r394552). This change is intended to provide a hook for a script running from udev once a span has been assigned ("registered") / unassigned ("unregistered") for its channels. The udev hook configures the span's channels with dahdi_cfg -S, and can then ask Asterisk to create ethe channels. See the scripts added to DAHDI-tools in 2.7.0. Review: https://reviewboard.asterisk.org/r/1598/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-08Add missing CR/LF to FakeMI stasis test AMI event.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396463 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-08Remove extra CR/LF from AMI event.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-08Blocked revisions 396441Walter Doekes
........ Consistent memory allocation by ast_bt_get_symbols. Always use ast_alloc/ast_free. This is handled differently in trunk (r391012). Review: https://reviewboard.asterisk.org/r/2580/ ........ Merged revisions 396427 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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