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2013-10-27Prevent CDR backends from unregistering while billing data is in flightMatthew Jordan
This patch makes it so that CDR backends cannot be unregistered while active CDR records exist. This helps to prevent billing data from being lost during restarts and shutdowns. Review: https://reviewboard.asterisk.org/r/2880/ ........ Merged revisions 402081 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-26chan_pjsip: Fix a crash when direct media is enabled and an ACK is received ↵Joshua Colp
after the channel is hung up. (closes issue ASTERISK-22731) Reported by: Kinsey Moore ........ Merged revisions 402064 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402065 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-26rtp_engine: fix rtp payloads copy and improve argument namesScott Griepentrog
In function ast_rtp_instance_early _bridge_make_compatible the use of instance 0/1 as arguments doesn't clearly communicate a direction that the copying of payloads from the source channel to the destination channel will occur, making it more probable to have the arguments to ast_rtp_codecs_payloads_copy() put in the reverse order. This patch renames the arguments with _dst and _src suffixes and corrects the copy direction. (closes issue ASTERISK-21464) Reported by: Kevin Stewart Review: https://reviewboard.asterisk.org/r/2894/ ........ Merged revisions 402000 from http://svn.asterisk.org/svn/asterisk/branches/1.8 Test shows rtpmap:119 being copied per this change, but is not in sip invite ........ Merged revisions 402042 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 402043 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-25pbx.c: fix confused match caller id that deleted exten still in hashScott Griepentrog
This fixes a bug where a zero length callerid match adjacent to a no match callerid extension entry would be deleted together, which then resulted in hashtable references to free'd memory. A third state of the matchcid value has been added to indicate match to any extension which allows enforcing comparison of matchcid on/off without errors. (closes issue AST-1235) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/2930/ ........ Merged revisions 401959 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 401960 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401961 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22Bridging: Fix orphaned bridge if neither of the joining channels can join.Richard Mudgett
The original issue noted that the bridge is orphaned when res_parking.so is not loaded and a call uses the dial kK flags. A similar issue happens when only one of the park flags is used. In this case you have the bridge with one or the other channel left in it. The channel and bridge will stay around until the channel hangs up. * Fixed the initial bridge channel push failure to act as if the channel were kicked out of the bridge. The bridge then decides if it needs to be dissolved. (closes issue ASTERISK-22629) Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/2928/ ........ Merged revisions 401424 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18Add channel lock protection around translation path setup.Richard Mudgett
Most callers of ast_channel_make_compatible() happen before the channels enter a two party bridge. With the new bridging framework, two party bridging technologies may also call ast_channel_make_compatible() when there is more than one thread involved with the two channels. * Added channel lock protection in set_format() and ast_channel_make_compatible_helper() when dealing with the channel's native formats while setting up a translation path. * Fixed best_src_fmt and best_dst_fmt usage consistency in ast_channel_make_compatible_helper(). The call to ast_translator_best_choice() got them backwards. * Updated some callers of ast_channel_make_compatible() and the function documentation. There is actually a difference between the two channels passed in. * Fixed the deadlock potential in res_fax.c dealing with ast_channel_make_compatible(). The deadlock potential was already there anyway because res_fax called ast_channel_make_compatible() with chan locked. (closes issue ASTERISK-22542) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2915/ ........ Merged revisions 401239 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18Tweak ast_bridge_depart() doxygen.Richard Mudgett
........ Merged revisions 401232 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401233 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18Remove the bit about requiring ast_bridge_depart() to be called before ↵Mark Michelson
ast_bridge_destroy(). ........ Merged revisions 401223 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18Clarify in ast_bridge_destroy() about how departable channels must be handled.Mark Michelson
........ Merged revisions 401212 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-11channel.h: whitespace changes.Richard Mudgett
........ Merged revisions 400854 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-07Miscellaneous stand alone comment cleanups.Richard Mudgett
........ Merged revisions 400661 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-04Add IPv6 Support To chan_iax2Michael L. Young
This patch adds IPv6 support to chan_iax2. Yay! (closes issue ASTERISK-22025) Patches: iax2-ipv6-v5-reviewboard.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2660/ ........ Merged revisions 400567 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-04ARI: Add subscription supportMatthew Jordan
This patch adds an /applications API to ARI, allowing explicit management of Stasis applications. * GET /applications - list current applications * GET /applications/{applicationName} - get details of a specific application * POST /applications/{applicationName}/subscription - explicitly subscribe to a channel, bridge or endpoint * DELETE /applications/{applicationName}/subscription - explicitly unsubscribe from a channel, bridge or endpoint Subscriptions work by a reference counting mechanism: if you subscript to an event source X number of times, you must unsubscribe X number of times to stop receiveing events for that event source. Review: https://reviewboard.asterisk.org/r/2862 (issue ASTERISK-22451) Reported by: Matt Jordan ........ Merged revisions 400522 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-03Fix assumption in bridge_native_rtp.c regarding number of participants in a ↵Mark Michelson
bridge. When a party leaves a bridge, there may be more participants in the bridge than expected. As such, it is important not to make assumptions regarding the list of channels in a bridge. This change makes it so that when a party leaves a native RTP bridge, we unbridge it and the party it was bridged with. Previously, the first and last channels in the list were unbridged since it was assumed that these were the two channels that had been bridged. As previously stated, a new party had been inserted into the bridge, so this logic did not work properly. (closes issue ASTERISK-22615) reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2899 ........ Merged revisions 400403 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-03Detect and use xsltCleanupGlobals when availableKinsey Moore
This introduces usage of an additional libxslt cleanup function, xsltCleanupGlobals, when the configure script detects that it is available. Early versions of the library did not include this function. (closes issue ASTERISK-22570) Reported by: Corey Farrell Patches: xsltCleanupGlobals.patch uploaded by Corey Farrell (License 5909) ........ Merged revisions 400384 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-03Cache string values of formats on ast_format_cap() to save processing.Mark Michelson
Channel snapshots have string representations of the channel's native formats. Prior to this change, the format strings were re-created on ever channel snapshot creation. Since channel native formats rarely change, this was very wasteful. Now, string representations of formats may optionally be stored on the ast_format_cap for cases where string representations may be requested frequently. When formats are altered, the string cache is marked as invalid. When strings are requested, the cache validity is checked. If the cache is valid, then the cached strings are copied. If the cache is invalid, then the string cache is rebuilt and copied, and the cache is marked as being valid again. Review: https://reviewboard.asterisk.org/r/2879 ........ Merged revisions 400356 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400363 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-02Multiple revisions 400318-400319Mark Michelson
........ r400318 | mmichelson | 2013-10-02 17:08:49 -0500 (Wed, 02 Oct 2013) | 12 lines Remove unnecessary waits from stasis. Since caches are updated on publisher threads, there is no need to wait for the cache updates to occur after a stasis message is published. In the case of chan_pjsip device state changes, this set of changes caused an improvement to performance. Review: https://reviewboard.asterisk.org/r/2890 ........ r400319 | mmichelson | 2013-10-02 17:10:54 -0500 (Wed, 02 Oct 2013) | 3 lines Remove svn:mergeinfo property. ........ Merged revisions 400318-400319 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-02Reduce channel snapshot creation and publishing by up to 50%.Joshua Colp
This change introduces the ability to stage channel snapshot creation and publishing by suppressing the implicit creation and publishing that some functions have. Once all operations are executed the staging is marked as done and a single snapshot is created and published. Review: https://reviewboard.asterisk.org/r/2889/ ........ Merged revisions 400265 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-02Allow specifying a channel to dial an extension and context in an ARI dial ↵Joshua Colp
operation. (issue ASTERISK-22625) Reported by: Scott Griepentrog ........ Merged revisions 400254 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-30Multiple revisions 399887,400138,400178,400180-400181David M. Lee
........ r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line Minor performance bump by not allocate manager variable struct if we don't need it ........ r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines 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/ ........ r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines 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/ ........ r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines 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/ ........ r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines 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/ ........ Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-27res_pjsip: crash when using localnet and external_signaling_address optionsKevin Harwell
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/ ........ Merged revisions 399990 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-27astobj2: Remove OBJ_CONTINUE support.Richard Mudgett
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/ ........ Merged revisions 399937 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-27Restore usefulness of the CEL Peer fieldKinsey Moore
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) ........ Merged revisions 399912 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-26pjsip: race condition in registrarKevin Harwell
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/ ........ Merged revisions 399897 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399898 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-17Change the "external_media_address" PJSIP endpoint option to "media_address".Mark Michelson
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 ........ Merged revisions 399283 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-17Fix doxygen to use correct units of features.conf options.Richard Mudgett
........ Merged revisions 399257 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-17Switch transferdigittimeout to be configured as seconds instead of milliseconds.Mark Michelson
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. ........ Merged revisions 399237 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13Restore Dial, Queue, and FollowMe 'I' option support.Richard Mudgett
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
2013-09-13Create more accurate Contact headers for dialogs when we are the UAS.Mark Michelson
(closes issue AST-1207) reported by John Bigelow Review: https://reviewboard.asterisk.org/r/2842 ........ Merged revisions 399083 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-09Fix DEBUG_THREADS when lock is acquired in __constructor__David M. Lee
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 ........ Merged revisions 398651 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-06astobj2: Add warn unused attribute to some functions.Richard Mudgett
* 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
2013-08-30Fix graceful shutdown crash.David M. Lee
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. ........ Merged revisions 398149 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398150 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-30Add a reloadable option for sorcery type objectsKevin Harwell
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/ ........ Merged revisions 398139 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-30optional_api: Fix linking problems between modules that export global symbolsDavid M. Lee
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/ ........ Merged revisions 397989 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-30ARI: Implement /recordings/stored API'sDavid M. Lee
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/ ........ Merged revisions 397985 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-28Match use of ast_free() with ast_calloc() and add some curly braces.Richard Mudgett
........ Merged revisions 397856 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-26bridging: Fix a livelock with local channel optimization.Richard Mudgett
Use a better means of waking up the bridge channel thread. ........ Merged revisions 397650 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-25Fix the config_options_testMatthew Jordan
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. ........ Merged revisions 397628 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23Add the bucket API.Joshua Colp
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
2013-08-23Fix a bug where the argc value was passed as no_doc when registering custom ↵Joshua Colp
sorcery types. This also adds a _nodoc equivalent. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397599 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23Handle DTMF and hold wrapup when a channel leaves the bridging system.Richard Mudgett
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
2013-08-23Fix memory corruption when trying to get "core show locks".Richard Mudgett
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
2013-08-23ARI: Correct error codes for bridge operationsDavid M. Lee
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
2013-08-23Add pass through support for Opus and VP8; Opus format attribute negotiationMatthew Jordan
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
2013-08-23Update config framework/sorcery with types/options without documentationMatthew Jordan
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
2013-08-23ARI: Music on Hold/Background Music for bridgesJonathan Rose
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
2013-08-22Add SayAlphaCase and similar functionality for AGIKinsey Moore
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
2013-08-22Update MOH start/stop routine doxygen.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Bridge API: Set a cause code on a channel when it is ejected from a bridge.Richard Mudgett
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
2013-08-22Massively clean up app_queue.Mark Michelson
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
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