asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	7d581b32e9	config: Allow options to register when documentation is unavailable. The config options framework is strict in that configuration options must be documented unless XML documentation support is not available. In practice this is useful as it ensures documentation exists however in off-nominal cases this can cause strange problems. If it is expected that a config option has a non-zero or non-empty default value but the config option documentation is unavailable this reasonable expectation will not be met. This can cause obscure crashes and weirdness depending on how the code handles it. This change tweaks the behavior to ensure that the config option is still allowed to register, apply default values, and be set when devmode is not enabled. If devmode is enabled then the option can NOT be set. This also does not remove the initial documentation error message that is output on load when registering the configuration option. ASTERISK-25725 #close Change-Id: Iec42fca6b35f31326c33fcdc25473f6fd7bc8af8 (cherry picked from commit `f22074e5d9`)	9 years ago
Mark Michelson	22eb1b48c0	res_pjsip_pubsub: Prevent crash from AMI command on freed subscription. A test recently uncovered that running an ill-timed AMI command to show inbound subscriptions could cause a crash since Asterisk will try to operate on a freed subscription. The fix for this is to remove the subscription tree from the list of subscriptions at the time that we are sending our final NOTIFY request out. This way, as the subscription is in the process of dying, it is inaccessible from AMI. Change-Id: Ic0239003d8d73e04c47c12dd2a7e23867e5b5b23 (cherry picked from commit `b073244c51`)	9 years ago
Richard Mudgett	826ff1d7a3	res/res_pjsip/presence_xml.c: Add missing 2nd call presence state case. ASTERISK-25712 #close Reported by: Richard Mudgett Change-Id: I70634df24f8c6c3a2c66c45af61d021e4999253f	9 years ago
Kevin Harwell	6e18a60a47	bridge_basic: don't cache xferfailsound during an attended transfer The xferfailsound was read from the channel at the beginning of the transfer, and that value is "cached" for the duration of the transfer. Therefore, changing the xferfailsound on the channel using the FEATURE() dialplan function does nothing once the transfer is under way. This makes it so the transfer code instead gets the xferfailsound configuration options from the channel when it is actually going to be used. This patch also fixes a potential memory leak of the props object as well as making sure the condition variable gets initialized before being destroyed. ASTERISK-25696 #close Change-Id: Ic726b0f54ef588bd9c9c67f4b0e4d787934f85e4	9 years ago
Joshua Colp	f63fb0e337	test_time: Provide a timeout when waiting. The test_timezone_watch unit test is written to expect a condition to be signaled when the inotify daemon thread runs. There exists a small window where the test_timezone_watch thread can signal the inotify daemon thread while it is not reading on the underlying file descriptor. If this occurs the test_timezone_watch thread will wait indefinitely for a signal that will never arrive. This change adds a timeout to the condition so it will return regardless after a period of time. Change-Id: Ifed981879df6de3d93acd3ee0a70f92546517390 (cherry picked from commit `c8499b8d5a`)	9 years ago
Joshua Colp	def98bb996	app: Queue hangup if channel is hung up during sub or macro execution. This issue was exposed when executing a connected line subroutine. When connected or redirected subroutines or macros are executed it is expected that the underlying applications and logic invoked are fast and do not consume frames. In practice this constraint is not enforced and if not adhered to will cause channels to continue when they shouldn't. This is because each caller of the connected or redirected logic does not check whether the channel has been hung up on return. As a result the the hung up channel continues. This change makes it so when the API to execute a subroutine or macro is invoked the channel is checked to determine if it has hung up. If it has then a hangup is queued again so the caller will see it and stop. ASTERISK-25690 #close Change-Id: I1f9a8ceb1487df0389f0d346ce0f6dcbcaf476ea	9 years ago
Kevin Harwell	bb29802615	pbx: Deadlock between contexts container and context_merge locks Recent changes (ASTERISK-25394 commit `2bd27d1222`) introduced the possibility of a deadlock. Due to the mentioned modifications ast_change_hints now needs to keep both merge/delete and state callbacks from occurring while it executes. Unfortunately, sometimes ast_change_hints can be called with the contexts container locked. When this happens it's possible for another thread to grab the context_merge_lock before the thread calling into ast_change_hints does and then try to obtain the contexts container lock. This of course causes a deadlock between the two threads. The thread calling into ast_change_hints waits for the other thread to release context_merge_lock and the other thread is waiting on that one to release the contexts container lock. Unfortunately, there is not a great way to fix this problem. When hints change, the subsequent state callbacks cannot run at the same time as a merge/delete, nor when the usual state callbacks do. This patch alleviates the problem by having those particular callbacks (the ones run after a hint change) occur in a serialized task. By moving the context_merge_lock to a task it can now safely be attempted or held without a deadlock occurring. ASTERISK-25640 #close Reported by: Krzysztof Trempala Change-Id: If2210ea241afd1585dc2594c16faff84579bf302	9 years ago
Mark Michelson	ca869878b4	PJSIP: Prevent deadlock due to dialog/transaction lock inversion. A deadlock was observed where the monitor thread was stuck, therefore resulting in no incoming SIP traffic being processed. The problem occurred when two 200 OK responses arrived in response to a terminating NOTIFY request sent from Asterisk. The first 200 OK was dispatched to a threadpool worker, who locked the corresponding transaction. The second 200 OK arrived, resulting in the monitor thread locking the dialog. At this point, the two threads are at odds, because the monitor thread attempts to lock the transaction, and the threadpool thread loops attempting to try to lock the dialog. In this case, the fix is to not have the monitor thread attempt to hold both the dialog and transaction locks at the same time. Instead, we release the dialog lock before attempting to lock the transaction. There have also been some debug messages added to the process in an attempt to make it more clear what is going on in the process. ASTERISK-25668 #close Reported by Mark Michelson Change-Id: I4db0705f1403737b4360e33a8e6276805d086d4a	10 years ago
Jonathan Rose	4e5aec3f0a	chan_sip: Add TCP/TLS keepalive to TCP/TLS server Adds the TCP Keep Alive option to TCP and TLS server sockets. Previously this option was only being set on session sockets. http://www.tldp.org/HOWTO/html_single/TCP-Keepalive-HOWTO/ According to the link above, the SO_KEEPALIVE option is useful for knowing when a TCP connected endpoint has severed communication without indicating it or has become unreachable for some reason. Without this patch, keep alive is not set on the socket listening for incoming TCP sessions and in Komatsu's report this resulted in the thread listening for TCP becoming stuck in a waiting state. ASTERISK-25364 #close Reported by: Hiroaki Komatsu Change-Id: I7ed7bcfa982b367dc64b4b73fbd962da49b9af36	10 years ago
Richard Mudgett	4d10ed67d0	PJSIP FAX: Fix T.38 automatic reject timer NULL channel pointer dereferences. When a caller calls a FAX number and then hangs up right after the call is answered then the T.38 re-INVITE automatic reject timer may still be running after the channel goes away. * Added session NULL channel checks on the code paths that get executed by t38_automatic_reject() to prevent a crash when the T.38 re-INVITE automatic reject timer expires. ASTERISK-25168 Reported by: Carl Fortin Change-Id: I07b6cd23815aedce5044f8f32543779e2f7a2403 (cherry picked from commit `8ea214aed7`)	10 years ago
Joshua Colp	a1e5bad564	Merge "sched.c: Make not return a sched id of 0." into certified/13.1	10 years ago
Joshua Colp	01d9ef510b	Merge topic 'ASTERISK-25476' into certified/13.1 * changes: Audit improper usage of scheduler exposed by `5c713fdf18`. (v13 additions) Audit improper usage of scheduler exposed by `5c713fdf18`.	10 years ago
Jonathan Rose	1ec791a3ba	Unset BRIDGEPEER when leaving a bridge Currently if a channel is transferred out of a bridge, the BRIDGEPEER variable (also BRIDGEPVTCALLID) remain set even once the channel is out of the bridge. This patch removes these variables when leaving the bridge. ASTERISK-25600 #close Reported by: Mark Michelson Change-Id: I753ead2fffbfc65427ed4e9244c7066610e546da	10 years ago
Richard Mudgett	a164f2ce7f	sched.c: Make not return a sched id of 0. According to the API doxygen a sched ID of 0 is valid. Unfortunately, 0 was never returned historically and several users incorrectly coded usage of the returned sched ID assuming that 0 was invalid. ASTERISK-25476 Change-Id: Ib19c7ebb44ec9fd393ef6646dea806d4f34e3a20	10 years ago
Richard Mudgett	a24db35ae3	Audit improper usage of scheduler exposed by `5c713fdf18`. (v13 additions) chan_sip.c: * Initialize mwi subscription scheduler ids earlier because of ASTOBJ to ao2 conversion. * Initialize register scheduler ids earlier because of ASTOBJ to ao2 conversion. chan_skinny.c: * Fix more scheduler usage for the valid 0 id value. ASTERISK-25476 Change-Id: If9f0e5d99638b2f9d102d1ebc9c5a14b2d706e95	10 years ago
Richard Mudgett	bea904e001	Audit improper usage of scheduler exposed by `5c713fdf18`. channels/chan_iax2.c: * Initialize struct chan_iax2_pvt scheduler ids earlier because of iax2_destroy_helper(). channels/chan_sip.c: channels/sip/config_parser.c: * Fix initialization of scheduler id struct members. Some off nominal paths had 0 as a scheduler id to be destroyed when it was never started. chan_skinny.c: * Fix some scheduler id comparisons that excluded the valid 0 id. channel.c: * Fix channel initialization of the video stream scheduler id. pbx_dundi.c: * Fix channel initialization of the packet retransmission scheduler id. ASTERISK-25476 Change-Id: I07a3449f728f671d326a22fcbd071f150ba2e8c8	10 years ago
Richard Mudgett	f5a6060707	res_sorcery_realtime.c: Fix crash from NULL sorcery object type. If the sorcery object type is not found a NULL is returned. Unfortunately, sorcery_realtime_filter_objectset() will crash after complaining about not finding the object type and saying to expect errors. * Use ao2_cleanup() instead of ao2_ref() to prevent the crash. ASTERISK-25165 Reported by Corey Farrell Change-Id: Ic3b64453ea3058cb68d5c26d97d4fe7b8eea2e97	10 years ago
Richard Mudgett	de43ae38b4	features: Fix crash when transferee hangs up during DTMF attended transfer. A crash happens with this sequence of steps: 1) Party A is connected to party B. 2) Party B starts a DTMF attended transfer. 3) Party A hangs up while party B is dialing party C. When party A hangs up the bridge that party A and party B are in is dissolved and party B is kicked out of the bridge. When party B finishes dialing party C he attempts to move to the new bridge with party C. Since party B is no longer in a bridge the attempted move dereferences a NULL bridge_channel pointer and crashes. * Made the hold(), unhold(), ringing(), and the bridge_move() functions tolerant of the channel not being in a bridge. The assertion that party B is always in a bridge is not true if the bridged peer of party B hangs up and dissolves the bridge. Being tolerant of not being in a bridge allows the peer hangup stimulus to be processed by the FSM. * Made the bridge_move() function return void since where the return value for a failed move was checked generated a FSM coding ERROR message for a normal off-nominal condition. * Eliminated most uses of RAII_VAR in bridge_basic.c. ASTERISK-25003 #close Reported by: Artem Volodin Change-Id: Ie2c1b14e5e647d4ea6de300bf56d69805d7bcada (cherry picked from commit `be1260a35f`)	10 years ago
Joshua Colp	89d40363ac	Merge "Confbridge: Add a user timeout option" into certified/13.1	10 years ago
Alec Davis	457d8dc124	app_queue: (try_calling): mutex 'qe->chan' freed more times than we've locked! commit `aae45acbd` (Mark Michelson 2015-04-15 10:38:02 -0500 6525) refer ASTERISK-24958 above commit removed ast_channel_lock(qe->chan); but failed to remove corresponding ast_channel_unlock(qe->chan); ASTERISK-25561 #close Reported Alec Davis Change-Id: Ie05f4e2d08912606178bf1fded57cc022c7a2e1a	10 years ago
Mark Michelson	c4751171a0	Confbridge: Add a user timeout option This option adds the ability to specify a timeout, in seconds, for a participant in a ConfBridge. When the user's timeout has been reached, the user is ejected from the conference with the CONFBRIDGE_RESULT channel variable set to "TIMEOUT". The rationale for this change is that there have been times where we have seen channels get "stuck" in ConfBridge because a network issue results in a SIP BYE not being received by Asterisk. While these channels can be hung up manually via CLI/AMI/ARI, adding some sort of automatic cleanup of the channels is a nice feature to have. ASTERISK-25549 #close Reported by Mark Michelson Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98	10 years ago
Mark Michelson	c0a7df1021	Taskprocessors: Increase high-water mark In practical tests, we have seen certain taskprocessors, specifically Stasis subscription taskprocessors, cross the recently-added high-water mark and emit a warning. This high-water mark warning is only intended to be emitted when things have tanked on the system and things are heading south quickly. In the practical tests, the Stasis taskprocessors sometimes had a max depth of 180 tasks in them, and Asterisk wasn't in any danger at all. As such, this ups the high-water mark to 500 tasks instead. It also redefines the SIP threadpool request denial number to be a multiple of the taskprocessor high-water mark. Change-Id: Ic8d3e9497452fecd768ac427bb6f58aa616eebce	10 years ago
Mark Michelson	2fc3267677	res_pjsip distributor: Don't send 503 response to responses. When the SIP threadpool is backed up with tasks, we send 503 responses to ensure that we don't try to overload ourselves. The problem is that we were not insuring that we were not trying to send a 503 to an incoming SIP response. This change makes it so that we only send the 503 on incoming requests. Change-Id: Ie2b418d89c0e453cc6c2b5c7d543651c981e1404	10 years ago
Matt Jordan	99c8d5c17f	Merge "Further fixes to improper usage of scheduler" into certified/13.1	10 years ago
Steve Davies	d760c21038	Further fixes to improper usage of scheduler When ASTERISK-25449 was closed, a number of scheduler issues mentioned in the comments were missed. These have since beed raised in ASTERISK-25476 and elsewhere. This patch attempts to collect all of the scheduler issues discovered so far and address them sensibly. ASTERISK-25476 #close Change-Id: I87a77d581e2e0d91d33b4b2fbff80f64a566d05b (cherry picked from commit `07583c2888`)	10 years ago
Mark Michelson	287cab1a53	res_pjsip: Deny requests when threadpool queue is backed up. We have observed situations where the SIP threadpool may become deadlocked. However, because incoming traffic is still arriving, the SIP threadpool's queue can continue to grow, eventually running the system out of memory. This change makes it so that incoming traffic gets rejected with a 503 response if the queue is backed up too much. Change-Id: I4e736d48a2ba79fd1f8056c0dcd330e38e6a3816	10 years ago
Joshua Colp	d073cb4b6d	threadpool: Handle worker thread transitioning to dead when going active. This change adds handling of dead worker threads when moving them to be active. When this happens the worker thread is removed from both the active and idle threads container. If no threads are able to be moved to active then the pool grows as configured. A unit test has also been added which thrashes the idle timeout and thread activation to exploit any race conditions between the two. ASTERISK-25546 #close Change-Id: I6c455f9a40de60d9e86458d447b548fb52ba1143	10 years ago
Jonathan Rose	b9713354dc	taskprocessor: Add high water mark warnings If a taskprocessor's queue grows large, this can indicate that there may be a problem with tasks not leaving the processor or else that the number of available task processors for a given type of task is too low. This patch makes it so that if a taskprocessor's task queue grows above 100 queued tasks that it will emit a warning message. Warning messages are emitted only once per task processor. ASTERISK-25518 #close Reported by: Jonathan Rose Change-Id: Ib1607c35d18c1d6a0575b3f0e3ff5d932fd6600c	10 years ago
Joshua Colp	ac9432fdb6	app_dial: Hold reference to calling channel formats when dialing outbound. Currently when requesting a channel the native formats of the calling channel are provided to the core for usage when dialing the outbound channel. This occurs without holding the channel lock or keeping a reference to the formats. This is problematic as the channel driver may end up changing the formats during this time. In the case of chan_sip this happens when an SDP negotiation completes. This change makes it so app_dial keeps a reference to the native formats of the calling channel which guarantees that they will remain valid for the period of time needed. ASTERISK-25172 #close Change-Id: I2f0a67bd0d5d14c3bdbaae552b4b1613a283f0db (cherry picked from commit `3b2b004d69`)	10 years ago
Matt Jordan	385e26efe2	main/dial: Protect access to the format_cap structure of the requesting channel When a dial attempt is made that involves a requesting channel, we previously were not: a) Protecting access to the native format capabilities structure on the requesting channel. That is inherently unsafe. b) Reference bumping the lifetime of the format capabilities structure. In both cases, something else could sneak in, blow away the format capabilities, and we'd be holding onto an invalid format_cap structure. When the newly created channel attempts to construct its format capabilities, things go poorly. This patch: a) Ensures that we get a reference to the native format capabilities while the requesting channel is locked b) Holds a reference to the native format capabilities during the creation of the new channel. ASTERISK-25522 #close Change-Id: I0bfb7ba8b9711f4158cbeaae96edf9626e88a54f	10 years ago
Mark Michelson	62799fe778	res_pjsip: Set threadpool max size default to 50. During a stress test of subscriptions, a huge blast of subscription-related traffic resulted in the threadpool expanding to a ridiculous number of threads. The balooning of threads resulted in an increase of memory, which led to a crash due to being out of memory. An easy fix for the particular test was to limit the size of the threadpool, thus reining in the amount of memory that would be used. It was decided that there really is no downside to having a non-infinite default value for the maximum size of the threadpool, so this change introduces 50 threads as the maximum threadpool size for the SIP threadpool. ASTERISK-25513 #close Reported by John Bigelow Change-Id: If0b9514f1d9b172540ce1a6e2f2ffa1f2b6119be	10 years ago
Matt Jordan	7b6fa02173	Merge "res_pjsip_pubsub: Prevent sending NOTIFY on destroyed dialog." into certified/13.1	10 years ago
Matt Jordan	1bbcaa833b	Merge "res_pjsip_pubsub: Ensure dialog lock balance." into certified/13.1	10 years ago
Matt Jordan	af98bbcb3f	Merge "res_pjsip_pubsub: Prevent crashes on final NOTIFY." into certified/13.1	10 years ago
Matt Jordan	834d3c54b7	Merge "res_pjsip_pubsub: Remove serializer when sending final NOTIFY." into certified/13.1	10 years ago
Matt Jordan	6efa8d44e8	Merge "res_pjsip_pubsub: Fix crash on destruction of empty subscription tree." into certified/13.1	10 years ago
Matt Jordan	2aea9b65a9	Merge "res_pjsip_pubsub: Solidify lifetime and ownership of objects." into certified/13.1	10 years ago
Matt Jordan	d701e51788	Merge "res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs." into certified/13.1	10 years ago
Kevin Harwell	6eda60936a	alembic: Bad down revision in add_default_from_user script The down revision wasn't set correct in the add_default_from_user script. This patch points it to the correct revision. Change-Id: Ied45786db265a1d4fb350ef0dd33b4d043c9a74d	10 years ago
Joshua Colp	3b9bd2d176	Merge "res_pjsip_outbound_registration: registration stops due to fatal 4xx response" into certified/13.1	10 years ago
Kevin Harwell	c425e26595	res_pjsip_outbound_registration: registration stops due to fatal 4xx response During outbound registration it is possible to receive a fatal (any permanent/ non-temporary 4xx, 5xx, 6xx) response from the registrar that is simply due to a problem with the registrar itself. Upon receiving the failure response Asterisk terminates outbound registration for the given endpoint. This patch adds an option, 'fatal_retry_interval', that when set continues outbound registration at the given interval up to 'max_retries' upon receiving a fatal response. ASTERISK-25485 #close Change-Id: Ibc2c7b47164ac89cc803433c0bbe7063bfa143a2	10 years ago
Joshua Colp	a0de15c8cf	Merge "Add missing failure checks to ast_str_set_va() callers." into certified/13.1	10 years ago
Joshua Colp	660ae8de8e	Merge "res_pjsip: Need to use the same serializer for a pjproject SIP transaction." into certified/13.1	10 years ago
Joshua Colp	08654131a4	Merge "Fix issue with AST_THREADSTORAGE_RAW when DEBUG_THREADLOCALS is enabled." into certified/13.1	10 years ago
Joshua Colp	74890feae5	Merge "Replace most uses of ast_register_atexit with ast_register_cleanup." into certified/13.1	10 years ago
Joshua Colp	61430bff17	Merge "res_pjsip: Move URI validation to use time." into certified/13.1	10 years ago
Mark Michelson	b95101aab0	format_cap: Detect vector allocation failures. A crash was seen on a system that ran out of memory due to Asterisk not checking for vector allocation failures in format_cap.c. With this change, if either of the AST_VECTOR_INIT calls fail, we will return a value indicating failure. Change-Id: Ieb9c59f39dfde6d11797a92b45e0cf8ac5722bc8	10 years ago
Mark Michelson	dd4d4e40e5	res_pjsip_pubsub: Prevent sending NOTIFY on destroyed dialog. A certain situation can result in our attempting to send a NOTIFY on a destroyed dialog. Say we attempt to send a NOTIFY to a subscriber, but that subscriber has dropped off the network. We end up retransmitting that NOTIFY until the appropriate SIP timer says to destroy the NOTIFY transaction. When the pjsip evsub code is told that the transaction has been terminated, it responds in kind by alerting us that the subscription has been terminated, destroying the subscription, and then removing its reference to the dialog, thus destroying the dialog. The problem is that when we get told that the subscription is being terminated, we detect that we have not sent a terminating NOTIFY request, so we queue up such a NOTIFY to be sent out. By the time that queued NOTIFY gets sent, the dialog has been destroyed, so attempting to send that NOTIFY can result in a crash. The fix being introduced here is actually a reintroduction of something the pubsub code used to employ. We hold a reference to the dialog and wait to decrement our reference to the dialog until our subscription tree object is destroyed. This way, we can send messages on the dialog even if the PJSIP evsub code wants to terminate earlier than we would like. In doing this, some NULL checks for subscription tree dialogs have been removed since NULL dialogs are no longer actually possible. Change-Id: I013f43cddd9408bb2a31b77f5db87a7972bfe1e5	10 years ago
Mark Michelson	bda0a24206	res_pjsip_pubsub: Ensure dialog lock balance. When sending a NOTIFY, we lock the dialog and then unlock the dialog when finished. A recent change made it so that the subscription tree's dialog pointer will be set NULL when sending the final NOTIFY request out. This means that when we attempt to unlock the dialog, we pass a NULL pointer to pjsip_dlg_dec_lock(). The result is that the dialog remains locked after we think we have unlocked it. When a response to the NOTIFY arrives, the monitor thread attempts to lock the dialog, but it cannot because we never released the dialog lock. This results in Asterisk being unable to process incoming SIP traffic any longer. The fix in this patch is to use a local pointer to save off the pointer value of the subscription tree's dialog when locking and unlocking the dialog. This way, if the subscription tree's dialog pointer is NULLed out, the local pointer will still have point to the proper place and the dialog lock will be unlocked as we expect. Change-Id: I7ddb3eaed7276cceb9a65daca701c3d5e728e63a	10 years ago
Mark Michelson	7a22fc27fb	res_pjsip_pubsub: Prevent crashes on final NOTIFY. The SIP dialog is removed from the subscription tree when the final NOTIFY is sent. However, after the final NOTIFY is sent, the persistence update function still attempts to access the cseq from the dialog, resulting in a crash. This fix removes the subscription persistence at the same time that the dialog is removed from the subscription tree. This way, there is no attempt to update persistence when the subscription is being destroyed. Change-Id: Ibb46977a6cef9c51dc95f40f43446e3d11eed5bb	10 years ago

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