5632 Commits (2c7ee034bcb8a4f8e18758aa6779b05428effd51)

Author	SHA1	Message	Date
Joshua Colp	def0cf7a74	Merge "Avoid setting maxfiles for a remote asterisk" into 14	9 years ago
Tzafrir Cohen	625e3e2e7b	Avoid setting maxfiles for a remote asterisk ... Setting maxfiles (maximum number of open files) has no practical effect on a remote asterisk (rasterisk, rasterisk -x). It has an ill effect of printing an extra message, which may be annoying in case of -x. ASTERISK-27105 #close Change-Id: Iaf9eb344e4b4b517df91b736b27ec55f6a6921a2	9 years ago
George Joseph	498768651e	http.c: Reduce log spam ... Messages like "fwrite() failed: Connection reset by peer" are no help whatsoever, especially since they can be caused simply by a client disconnecting. * Make those WARNINGs DEBUGs. * Check the return of the headers fprintf. Change-Id: I17bd5f3621514152a7b2b263c801324c5e96568b	9 years ago
Richard Mudgett	f675622707	json.c: Add backtrace log to find 'Invalid UTF-8 string' errors ... Change-Id: I9020ff9f2b3749904317c0c173f47a1bbed6f929	9 years ago
Joshua Colp	4ce81fb759	Merge "core: Fix segfault when invoking 'data get' CLI command" into 14	9 years ago
Sean Bright	ebacfc8618	core: Fix segfault when invoking 'data get' CLI command ... Invoking 'data get /asterisk/core/channeltypes' caused a crash because of an assumption of a tech's capabilities to be non-NULL. The 'Surrogate' tech, however, does have a NULL capabilities member, resulting in a crash. ASTERISK-27108 #close Change-Id: I2fbe7715681f43d5565d1e1599269468c26b0e0a	9 years ago
Corey Farrell	584f969020	channel: Clear channel flag in error branch. ... Clear channel flag AST_FLAG_END_DTMF_ONLY in ast_waitfordigit_full when ast_read returns NULL. ASTERISK-27100 #close Change-Id: Id3039e9a4e74e0cb359f636c9fd0c9740ebf7d9d	9 years ago
Jenkins2	d1344a9caf	Merge "bridge: stuck channel(s) after failed attended transfer" into 14	9 years ago
Kevin Harwell	e733df06cb	core_local: local channel data not being properly unref'ed and unlocked ... In an earlier version of Asterisk a local channel [un]lock all functions were added in order to keep a crash from occurring when a channel hung up too early during an attended transfer. Unfortunately, when a transfer failure occurs and depending on the timing, the local channels sometime do not get properly unlocked and deref'ed after being locked and ref'ed. This happens because the underlying local channel structure gets NULLed out before unlocking. This patch reworks those [un]lock functions and makes sure the values that get locked and ref'ed later get unlocked and deref'ed. ASTERISK-27074 #close Change-Id: Ice96653e29bd9d6674ed5f95feb6b448ab148b09	9 years ago
Kevin Harwell	6a2fe083ce	bridge: stuck channel(s) after failed attended transfer ... If an attended transfer failed it was possible for some of the channels involved to get "stuck" because Asterisk was not hanging up the transfer target. This patch ensures Asterisk hangs up the transfer target when an attended transfer failure occurs. ASTERISK-27075 #close Change-Id: I98a6ecd92d3461ab98c36f0d9451d23adaf3e5f9	9 years ago
Jenkins2	82c37b5b51	Merge "res_stasis: Plug reference leak on stolen channels" into 14	9 years ago
George Joseph	4a66babfe0	res_stasis: Plug reference leak on stolen channels ... When a stasis channel is stolen by another app, the control structure is unreffed but never unlinked from the app_controls container. This causes the channel reference to leak. Added OBJ_UNLINK to the callback in channel_stolen_cb. Also added some additional channel lifecycle debug messages to channel.c. ASTERISK-27059 #close Repoorted-by: George Joseph Change-Id: Ib820936cd49453f20156971785e7f4f182c56e14	9 years ago
Frederic LE FOLL	053d92b0ba	Core/PBX: Deadlock between dialplan execution and application unregistration. ... Not easy to reproduce, but we have noticed deadlocks when unloading a module while dialplan is handling a request. The deadlock is between : 1) Dialplan execution: pbx_extension_helper() first taking conlock, then pbx_findapp() [when called] asking for lock on apps list. 2) Application unregistration: ast_unregister_application() first taking lock on apps list, then unreference_cached_app() [when called] asking for conlock. As a protection, I suggest to modify ast_unregister_application(), so that it anticipates the need of conlock, before taking the lock on apps list. The side effect is a longer unavailability of conlock when unregistering an application. ASTERISK-27041 Change-Id: I0db0f1eb320da6a5758cce3a47d765be1face8e2	9 years ago
Jenkins2	f2e0160a71	Merge "channel: Fix reference counting in ast_channel_suppress." into 14	9 years ago
Jenkins2	b0d3cc079f	Merge "bridge: Add a deferred queue." into 14	9 years ago
Joshua Colp	8485d2fc36	channel: Fix reference counting in ast_channel_suppress. ... The ast_channel_suppress function wrongly decremented the reference count of the underlying structure used to keep track of what should be suppressed on a channel if the function was called multiple times on the same channel. This change cleans up the reference counting a bit so this no longer occurs. ASTERISK-27016 Change-Id: I2eed4077cb4916e6626f9f120b63b963acc5c136	9 years ago
Joshua Colp	bbc2d6f214	Merge "BuildSystem: Add patches to allow building with recent LibreSSL" into 14	9 years ago
Joshua Colp	0c6c258a7e	bridge: Add a deferred queue. ... This change adds a deferred queue to bridging. If a bridge technology determines that a frame can not be written and should be deferred it can indicate back to bridging to do so. Bridging will then requeue any deferred frames upon a new channel joining the bridge. This change has been leveraged for T.38 request negotiate control frames. Without the deferred queue there is a race condition between the bridge receiving the T.38 request negotiate and the second channel joining and being in the bridge. If the channel is not yet in the bridge then the T.38 negotiation fails. A unit test has also been added that confirms that a T.38 request negotiate control frame is deferred when no other channel is in the bridge and that it is requeued when a new channel joins the bridge. ASTERISK-26923 Change-Id: Ie05b08523f399eae579130f4a5f562a344d2e415	9 years ago
Guido Falsi	e1366359f1	BuildSystem: Add patches to allow building with recent LibreSSL ... Add some #if defined checks which allow building against LibreSSL. These patchess come from OpenBSD ports: https://cvsweb.openbsd.org/cgi-bin/cvsweb/ports/telephony/asterisk/patches/ ASTERISK-27043 #close Reported by: OpenBSD ports Change-Id: I2f6c08a5840b85ad4d2b75370b947ddde7a9a572	9 years ago
Guido Falsi	d3cf5492dd	BuildSystem: Fix build on FreeBSD due to missing crypt.h ... FreeBSD does not include a crypt.h include file. Definitions for crypt() and crypt_r() are in unistd.h ASTERISK-27042 #close Change-Id: Ib307ee5e384870c6af50efa89fb73722dd0c3a7e	9 years ago
Joshua Colp	4ab80f6aa0	Merge "channel: ast_write frame wrongly freed after call to audiohooks" into 14	9 years ago
Jenkins2	e3d290c957	Merge "format: Reintroduce smoother flags" into 14	9 years ago
Kevin Harwell	ecc311f12d	channel: ast_write frame wrongly freed after call to audiohooks ... ASTERISK-26419 introduced a bug when calling ast_audiohook_write_list in ast_write. It would free the frame given to ast_write if the frame returned by ast_audiohook_write_list was different than the given one. The frame give to ast_write should never be freed within that function. It is the caller's resposibility to free the frame after writing (or when it its done with it). By freeing it within ast_write this of course led to some memory corruption problems. This patch makes it so the frame given to ast_write is no longer freed within the function. The frame returned by ast_audiohook_write_list is now subsequently used in ast_write and is freed later. It is freed either after translate if the frame returned by translate is different, or near the end of ast_write prior to function exit. ASTERISK-26973 #close Change-Id: I463d4ac3b736ced95de986ee74a489c7c7ab103b	9 years ago
Sean Bright	7bdaed63fd	pbx_builtin: Properly handle hangup during Background ... Before this patch, when a user hung up during a Background, we would stuff 0xff into a char and attempt a dialplan lookup of it. This caused problems for some realtime engines which interpreted the value as the beginning of an invalid UTF-8 sequence. ASTERISK-19291 #close Reported by: Andrew Nowrot Change-Id: I8ca6da93252d61c76ebdb46a4aa65e73ca985358	9 years ago
Joshua Colp	b86c8803fe	channel / app_meetme: Fix parentheses. ... ASTERISK-27025 Change-Id: Id736b0aa4ec6b6b0f04663d64fa8d151f81fdbed	9 years ago
Sean Bright	ca2b3584b2	format: Reintroduce smoother flags ... In review 4843 (ASTERISK-24858), we added a hack that forced a smoother creation when sending signed linear so that the byte order was adjusted during transmission. This was needed because smoother flags were lost during the new format work that was done in Asterisk 13. Rather than rolling that same hack into res_rtp_multicast, re-introduce smoother flags so that formats can dictate their own options. Change-Id: I77b835fba0e539c6ce50014a984766f63cab2c16	9 years ago
Joshua Colp	61e4fc5f2d	manager: Clear the flag on the other channel. ... During the channel flag audit an incorrect change was done. The flag should be cleared on the second channel. ASTERISK-26469 Change-Id: I770c5a389550a2fb5a6ade942fccbb2e1d9199c8	9 years ago
Jenkins2	4e757a4ca0	Merge "asterisk: Audit locking of channel when manipulating flags." into 14	9 years ago
George Joseph	61b9b3fd11	unittests: Add a unit test that causes a SEGV and... ... ...that can only be run by explicitly calling it with 'test execute category /DO_NOT_RUN/ name RAISE_SEGV' This allows us to more easily test CI and debugging tools that should do certain things when asterisk coredumps. To allow this a new member was added to the ast_test_info structure named 'explicit_only'. If set by a test, the test will be skipped during a 'test execute all' or 'test execute category ...'. Change-Id: Ia3a11856aae4887df9a02b6b081cc777b36eb6ed	9 years ago
Joshua Colp	ede77acaaf	asterisk: Audit locking of channel when manipulating flags. ... When manipulating flags on a channel the channel has to be locked to guarantee that nothing else is also manipulating the flags. This change introduces locking where necessary to guarantee this. It also adds helper functions that manipulate channel flags and lock to reduce repeated code. ASTERISK-26789 Change-Id: I489280662dba0f4c50981bfc5b5a7073fef2db10	9 years ago
Jenkins2	a0e3da1f71	Merge "logger: Added logger_queue_limit to the configuration options." into 14	9 years ago
Joshua Colp	d6a9beaab1	Merge "Prevent Undefined Capath Crash" into 14	9 years ago
Joshua Colp	ecd371846e	tcptls: Improve error messages for TLS connections. ... This change uses the functions provided by OpenSSL to query and better construct error messages for situations where the connection encounters a problem. ASTERISK-26606 Change-Id: I7ae40ce88c0dc4e185c4df1ceb3a6ccc198f075b	9 years ago
Joshua Elson	ee019a5f77	Prevent Undefined Capath Crash ... It is possible to initialize a valid config without a capath or cafile definition. This will cause a crash on a reload. This fix ensures capath is always allocated. ASTERISK-26983 #close Change-Id: I63ff715d9d9023427543a5b8a4ba7b0d82533c12	9 years ago
George Joseph	944f9435bc	logger: Added logger_queue_limit to the configuration options. ... All log messages go to a queue serviced by a single thread which does all the IO. This setting controls how big that queue can get (and therefore how much memory is allocated) before new messages are discarded. The default is 1000. Should something go bezerk and log tons of messages in a tight loop, this will prevent memory escalation. When the limit is reached, a WARNING is logged to that effect and messages are discarded until the queue is empty again. At that time another WARNING will be logged with the count of discarded messages. There's no "low water mark" for this queue because the logger thread empties the entire queue and processes it in 1 batch before going back and waiting on the queue again. Implementing a low water mark would mean additional locking as the thread processes each message and it's not worth it. A "test" was added to test_logger.c but since the outcome is non-deterministic, it's really just a cli command, not a unit test. Change-Id: Ib4520c95e1ca5325dbf584c7989ce391649836d1	9 years ago
Joshua Colp	6b9ed2574b	Merge "netsock2.c: Made get/set addr port avoid potential uninitialized memory." into 14	9 years ago
Joshua Colp	eb38be715a	Merge "bridge: Fix returning to dialplan when executing Bridge() from AMI." into 14	9 years ago
Richard Mudgett	fe06758718	netsock2.c: Made get/set addr port avoid potential uninitialized memory. ... Change-Id: I532052bd7cd95a4b3565485fc01e2a1ea07ee647	9 years ago
George Joseph	e9d563c4db	app_confbridge: Fix reference to cfg in menu_template_handler ... menu_template_handler wasn't properly accounting for the fact that it might be called both during a load/reload (which isn't really valid but not prevented) and by a dialplan function. In both cases it was attempting to use the "pending" config which wasn't valid in the latter case. aco_process_config is also partly to blame because it wasn't properly cleaning "pending" up when a reload was done and no changes were made. Both of these contributed to a crash if CONFBRIDGE(menu,template) was called in a dialplan after a reload. * aco_process_config now sets info->internal->pending to NULL after it unrefs it although this isn't strictly necessary in the context of this fix. * menu_template_handler now uses the "current" config and silently ignores any attempt to be called as a result of someone uses the "template" parameter in the conf file. Luckily there's no other place in the codebase where aco_pending_config is used outside of aco_process_config. ASTERISK-25506 #close Reported-by: Frederic LE FOLL Change-Id: Ib349a17d3d088f092480b19addd7122fcaac21a7	9 years ago
Joshua Colp	8cbdd0a5b8	bridge: Fix returning to dialplan when executing Bridge() from AMI. ... When using the Bridge AMI action on the same channel multiple times it was possible for the channel to return to the wrong location in the dialplan if the other party hung up. This happened because the priority of the channel was not preserved across each action invocation and it would fail to move on to the next priority in other cases. This change makes it so that the priority of a channel is preserved when taking control of it from another thread and it is incremented as appropriate such that the priority reflects where the channel should next be executed in the dialplan, not where it may or may not currently be. The Bridge AMI action was also changed to ensure that it too starts the channels at the next location in the dialplan. ASTERISK-24529 Change-Id: I52406669cf64208aef7252a65b63ade31fbf7a5a	9 years ago
Jenkins2	a5e79f71ee	Merge "frame: Better handle interpolated frames." into 14	9 years ago
Joshua Colp	4cdf937a2e	frame: Better handle interpolated frames. ... Interpolated frames are frames which contain a number of samples but have no actual data. Audiohooks did not handle this case when translating an incoming frame into signed linear. It assumed that a frame would always contain media when it may not. If this occurs audiohooks will now immediately return and not act on the frame. As well for users of ast_trans_frameout the function has been changed to be a bit more sane and ensure that the data pointer on a frame is set to NULL if no data is actually on the frame. This allows the various spots in Asterisk that check for an interpolated frame based on the presence of a data pointer to work as expected. ASTERISK-26926 Change-Id: I7fa22f631fa28d540722ed789ce28e84c7f8662b	9 years ago
Sean Bright	4cb1458245	cleanup: Fix fread() and fwrite() error handling ... Cleaned up some of the incorrect uses of fread() and fwrite(), mostly in the format modules. Neither of these functions will ever return a value less than 0, which we were checking for in some cases. I've introduced a fair amount of duplication in the format modules, but I plan to change how format modules work internally in a subsequent patch set, so this is simply a stop-gap. Change-Id: I8ca1cd47c20b2c0b72088bd13b9046f6977aa872	9 years ago
Sean Bright	d757a70156	core: Use eventfd for alert pipes on Linux when possible ... The primary win of switching to eventfd when possible is that it only uses a single file descriptor while pipe() will use two. This means for each bridge channel we're reducing the number of required file descriptors by 1, and - if you're using timerfd - we also now have 1 less file descriptor per Asterisk channel. The API is not ideal (passing int arrays), but this is the cleanest approach I could come up with to maintain API/ABI. I've also removed what I believe to be an erroneous code block that checked the non-blocking flag on the pipe ends for each read. If the file descriptor is 'losing' its non-blocking mode, it is because of a bug somewhere else in our code. In my testing I haven't seen any measurable difference in performance. Change-Id: Iff0fb1573e7f7a187d5211ddc60aa8f3da3edb1d	9 years ago
George Joseph	a40f2c8246	Merge "pbx: Use same thread if AST_OUTGOING_WAIT_COMPLETE specified" into 14	9 years ago
Sean Bright	6bbccb0dfd	pbx: Use same thread if AST_OUTGOING_WAIT_COMPLETE specified ... Both ast_pbx_outgoing_app() and ast_pbx_outgoing_exten() cause the core to spawn a new thread to perform the dial. When AST_OUTGOING_WAIT_COMPLETE is passed to these functions, the calling thread will be blocked until the newly created channel has been hung up. After this patch, we run the dial on the current thread rather than spawning a new one. The only in-tree code that passes AST_OUTGOING_WAIT_COMPLETE is pbx_spool, so you should see reduced thread usage if you are using .call files. Change-Id: I512735d243f0a9da2bcc128f7a96dece71f2d913	9 years ago
Richard Mudgett	3adb547069	rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes. ... The struct ast_rtp_instance has historically been indirectly protected from reentrancy issues by the channel lock because early channel drivers held the lock for really long times. Holding the channel lock for such a long time has caused many deadlock problems in the past. Along comes chan_pjsip/res_pjsip which doesn't necessarily hold the channel lock because sometimes there may not be an associated channel created yet or the channel pointer isn't available. In the case of ASTERISK-26835 a pjsip serializer thread was processing a message's SDP body while another thread was reading a RTP packet from the socket. Both threads wound up changing the rtp->rtcp->local_addr_str string and interfering with each other. The classic reentrancy problem resulted in a crash. In the case of ASTERISK-26853 a pjsip serializer thread was processing a message's SDP body while another thread was reading a RTP packet from the socket. Both threads wound up processing ICE candidates in PJPROJECT and interfering with each other. The classic reentrancy problem resulted in a crash. * rtp_engine.c: Make the ast_rtp_instance_xxx() calls lock the RTP instance struct. * rtp_engine.c: Make ICE and DTLS wrapper functions to lock the RTP instance struct for the API call. * res_rtp_asterisk.c: Lock the RTP instance to prevent a reentrancy problem with rtp->rtcp->local_addr_str in the scheduler thread running ast_rtcp_write(). * res_rtp_asterisk.c: Avoid deadlock when local RTP bridging in bridge_p2p_rtp_write() because there are two RTP instance structs involved. * res_rtp_asterisk.c: Avoid deadlock when trying to stop scheduler callbacks. We cannot hold the instance lock when trying to stop a scheduler callback. * res_rtp_asterisk.c: Remove the lock in struct dtls_details and use the struct ast_rtp_instance ao2 object lock instead. The lock was used to synchronize two threads to prevent a race condition between starting and stopping a timeout timer. The race condition is no longer present between dtls_perform_handshake() and __rtp_recvfrom() because the instance lock prevents these functions from overlapping each other with regards to the timeout timer. * res_rtp_asterisk.c: Remove the lock in struct ast_rtp and use the struct ast_rtp_instance ao2 object lock instead. The lock was used to synchronize two threads using a condition signal to know when TURN negotiations complete. * res_rtp_asterisk.c: Avoid deadlock when trying to stop the TURN ioqueue_worker_thread(). We cannot hold the instance lock when trying to create or shut down the worker thread without a risk of deadlock. This patch exposed a race condition between a PJSIP serializer thread setting up an ICE session in ice_create() and another thread reading RTP packets. * res_rtp_asterisk.c:ice_create(): Set the new rtp->ice pointer after we have re-locked the RTP instance to prevent the other thread from trying to process ICE packets on an incomplete ICE session setup. A similar race condition is between a PJSIP serializer thread resetting up an ICE session in ice_create() and the timer_worker_thread() processing the completion of the previous ICE session. * res_rtp_asterisk.c:ast_rtp_on_ice_complete(): Protect against an uninitialized/null remote_address after calling update_address_with_ice_candidate(). * res_rtp_asterisk.c: Eliminate the chance of ice_reset_session() destroying and setting the rtp->ice pointer to NULL while other threads are using it by adding an ao2 wrapper around the PJPROJECT ice pointer. Now when we have to unlock the RTP instance object to call a PJPROJECT ICE function we will hold a ref to the wrapper. Also added some rtp->ice NULL checks after we relock the RTP instance and have to do something with the ICE structure. ASTERISK-26835 #close ASTERISK-26853 #close Change-Id: I780b39ec935dcefcce880d50c1a7261744f1d1b4	9 years ago
George Joseph	cc668bd522	modules: change module LOAD_FAILUREs to LOAD_DECLINES ... In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting if a module can't be loaded. If the user wishes to retain the FAILURE behavior for a specific module, they can use the "require" or "preload-require" keyword in modules.conf. A new API was added to logger: ast_is_logger_initialized(). This allows asterisk.c/check_init() to print to the error log once the logger subsystem is ready instead of just to stdout. If something does fail before the logger is initialized, we now print to stderr instead of stdout. Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25	9 years ago
Joshua Colp	c4bbf0c0db	Merge changes from topic 'ASTERISK-26890' into 14 ... * changes: stun.c: Fix ast_stun_request() erratic timeout. sorcery.c: Speed up ast_sorcery_retrieve_by_id() res_pjsip: Fix pointer use after unref.	9 years ago
Richard Mudgett	3408561eb7	stun.c: Fix ast_stun_request() erratic timeout. ... If ast_stun_request() receives packets other than a STUN response then we could conceivably never exit if we continue to receive packets with less than three seconds between them. * Fix poll timeout to keep track of the time when we sent the STUN request. We will now send a STUN request every three seconds regardless of how many other packets we receive while waiting for a response until we have completed three STUN request transmission cycles. Change-Id: Ib606cb08585e06eb50877f67b8d3bd385a85c266	9 years ago