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Author	SHA1	Message	Date
Joshua Colp	771757e338	Merge "translate: Fix transcoding while different in frame size." into 13	10 years ago
Joshua Colp	bc856bda64	Merge "app_dial.c: Factor out a connected line update routine." into 13	10 years ago
Matt Jordan	9e7c554089	Merge "res/res_stasis: Fix accidental subscription to 'all' bridge topic" into 13	10 years ago
Joshua Colp	7a5ce4ad5a	Merge "app_dial.c: Remove some no-op code." into 13	10 years ago
Matt Jordan	90165e306d	res/res_stasis: Fix accidental subscription to 'all' bridge topic ... When b99a705262 was merged, subscribing to a NULL bridge will now cause app_subscribe_bridge to implicitly subscribe to all bridges. Unfortunately, the res_stasis control loop did not check that a bridge changing on a channel's control object was actually also non-NULL. As a result, app_subscribe_bridge will be called with a NULL bridge when a channel leaves a bridge. This causes a new subscription to be made to the bridge. If an application has also subscribed to the bridge, the application will now have two subscriptions: (1) The explicit one created by the app (2) The implicit one accidentally created by the control structure As a result, the 'BridgeDestroyed' event can be sent multiple times. This patch corrects the control loop such that it only subscribes an application to a new bridge if the bridge pointer is non-NULL. ASTERISK-24870 Change-Id: I3510e55f6bc36517c10597ead857b964463c9f4f	10 years ago
Scott Griepentrog	e1223ff6db	Scripts: check file versions of Asterisk and dependencies ... To help in diagnosing mismatched modules and libraries, this script scans for version, repository, and source information and reports what is found. ASTERISK-25376 #close Reported by: Ashley Sanders Change-Id: Ib0642d0fb96712476f59760d6d137a24633fe2d6	10 years ago
Richard Mudgett	14481d9aa0	app_dial.c: Factor out a connected line update routine. ... Replace inlined code with update_connected_line_from_peer(). ASTERISK-25423 Reported by: John Hardin Change-Id: Ia14f18def417645cd7fb453e1bdac682630a5091	10 years ago
Richard Mudgett	bbeda190c3	app_dial.c: Remove some no-op code. ... Change-Id: Ice1884a94315d3cb7e3bbd47a9fba76a27276c54	10 years ago
Joshua Colp	41f856e5a2	Merge "logger: Prevent duplicate dynamic channels from being added." into 13	10 years ago
Mark Michelson	f050fa76eb	logger: Prevent duplicate dynamic channels from being added. ... There was a problem observed where the "logger add channel" CLI command would allow for a channel with the same name to be added multiple times. This would result in each message being written out to the same file multiple times. The problem was due to the difference in how logger channel filenames are stored versus the format they are allowed to be presented when they are added. For instance, if adding the logger channel "foo" through the CLI, the result would be a logger channel with the file name /var/log/asterisk/foo being stored. So when trying to add another "foo" channel, "foo" would not match "/var/log/asterisk/foo" so we'd happily add the duplicate channel. The fix presented here is to introduce two new methods in the logger code: * make_filename(): given a logger channel name, this creates the filename for that logger channel. * find_logchannel(): given a logger channel name, this calls make_filename() and then traverses the list of logchannels in order to find a match. This change has made use of make_filename() and find_logchannel() throughout to more consistently behave. ASTERISK-25305 #close Reported by Mark Michelson Change-Id: I892d52954d6007d8bc453c3cbdd9235dec9c4a36	10 years ago
Mark Michelson	629458d349	Do not swallow frames on channels leaving bridges. ... When leaving a bridge, indications on a channel could be swallowed by the internal indication logic because it appears that the channel is on its way to be hung up anyway. One such situation where this is detrimental is when channels on hold are redirected out of a bridge. The AST_CONTROL_UNHOLD indication from the bridging code is swallowed, leaving the channel in question to still appear to be on hold. The fix here is to modify the logic inside ast_indicate_data() to not drop the indication if the channel is simply leaving a bridge. This way, channels on hold redirected out of a bridge revert to their expected "in use" state after the redirection. ASTERISK-25418 #close Reported by Mark Michelson Change-Id: If6115204dfa0551c050974ee138fabd15f978949	10 years ago
Matt Jordan	0d95610ae2	Merge "ARI: Add events for Contact and Peer Status changes" into 13	10 years ago
Richard Mudgett	5f15cd93f0	app_page.c: Fix crash when forwarding with a predial handler. ... Page uses the async method of dialing with the dial API. When a call gets forwarded there is no calling channel available. If the predial handler was set then the calling channel could not be put into auto-service for the forwarded call because it doesn't exist. A crash is the result. * Moved the callee predial parameter string processing to before the string is passed to the dial API rather than having the dial API do it. There are a few benefits do doing this. The first is the predial parameter string processing doesn't need to be done for each channel called by the dial API. The second is in async mode and the forwarded channel is to have the predial handler executed on it then the non-existent calling channel does not need to be present to process the predial parameter string. * Don't start auto-service on a non-existent calling channel to execute the predial handler when the dial API is in async mode and forwarding a call. ASTERISK-25384 #close Reported by: Chet Stevens Change-Id: If53892b286d29f6cf955e2545b03dcffa2610981	10 years ago
Matt Jordan	b50e372394	ARI: Add events for Contact and Peer Status changes ... This patch adds support for receiving events regarding Peer status changes and Contact status changes. This is particularly useful in scenarios where we are subscribed to all endpoints and channels, where we often want to know more about the state of channel technology specific items than a single endpoint's state. ASTERISK-24870 Change-Id: I6137459cdc25ce27efc134ad58abf065653da4e9	10 years ago
Matt Jordan	3502c0431d	res/res_stasis_device_state: Allow for subscribing to 'all' device state ... This patch adds support for subscribing to all device state changes. This is done either by subscribing to an empty device, e.g., 'eventSource=deviceState:', or by the WebSocket connection specifying that it wants all state in the system. ASTERISK-24870 Change-Id: I9cfeca1c9e2231bd7ea73e45919111d44d2eda32	10 years ago
Matt Jordan	4c9f613309	ARI: Add the ability to subscribe to all events ... This patch adds the ability to subscribe to all events. There are two possible ways to accomplish this: (1) On initial WebSocket connection. This patch adds a new query parameter, 'subscribeAll'. If present and True, Asterisk will subscribe the applications to all ARI events. (2) Via the applications resource. When subscribing in this manner, an ARI client should merely specify a blank resource name, i.e., 'channels:' instead of 'channels:12354'. This will subscribe the application to all resources of the 'channels' type. ASTERISK-24870 #close Change-Id: I4a943b4db24442cf28bc64b24bfd541249790ad6	10 years ago
Elazar Broad	ec514ad64d	core/logging: Fix logging to more than one syslog channel ... Currently, Asterisk will log to the last configured syslog channel in logger.conf. This is due to the fact that the final call to openlog() supersedes all of the previous calls. This commit removes the call to openlog() and passes the facility to ast_log_vsyslog(), along with utilizing the LOG_MAKEPRI macro to ensure that the message is routed to the correct facility and with the correct priority. ASTERISK-25407 #close Reported by: Elazar Broad Tested by: Elazar Broad Change-Id: Ie2a2416bc00cce1b04e99ef40917c2011953ddd2	10 years ago
Matt Jordan	0ce661f0c0	Merge "app_record: RECORDED_FILE variable not being populated" into 13	10 years ago
Joshua Colp	a45abbf3ce	Merge "pbx: Update device and presence state when changing a hint extension." into 13	10 years ago
Kevin Harwell	aeddee39fb	app_record: RECORDED_FILE variable not being populated ... The RECORDED_FILE variable is empty unless a '%d' is specified in the filename. This patch makes it so the variable is always set to the filename. ASTERISK-25410 #close Change-Id: I4ec826d8eb582ae2ad184e717be8668b74d37653	10 years ago
Matt Jordan	960f00939a	Merge "main/config_options: Check for existance of internal object before derefing" into 13	10 years ago
Matt Jordan	efbd0f5a97	Merge "app_queue: AgentComplete event has wrong reason" into 13	10 years ago
Matt Jordan	c7beb33ebb	Merge "app_queue: Crash when transferring" into 13	10 years ago
Matt Jordan	16f1598b31	Merge "CHAOS: res_pjsip_diversion avoid crash if allocation fails" into 13	10 years ago
Joshua Colp	c02f07b8c4	Merge "chan_sip: Fix From header truncation for extremely long CALLERID(name)." into 13	10 years ago
Joshua Colp	578429a54d	Merge "CHAOS: avoid crash if string create fails" into 13	10 years ago
Joshua Colp	2bd27d1222	pbx: Update device and presence state when changing a hint extension. ... When changing a hint extension without removing the hint first the device state and presence state is not updated. This causes the state of the hint to be that of the previous extension and not the current one. This state is kept until a state change occurs as a result of something (presence state change, device state change). This change updates the hint with the current device and presence state of the new extension when it is changed. Any state callbacks which may have been added before the hint extension is changed are also informed of the new device and presence state if either have changed. ASTERISK-25394 #close Change-Id: If268f1110290e502c73dd289c9e7e7b27bc8432f	10 years ago
Scott Griepentrog	c94f46080f	CHAOS: avoid crash if string create fails ... Validate string buffer allocation before using them. ASTERISK-25323 Change-Id: Ib9c338bdc1e53fb8b81366f0b39482b83ef56ce0	10 years ago
Walter Doekes	b59c4d82b5	chan_sip: Fix From header truncation for extremely long CALLERID(name). ... The CALLERID(num) and CALLERID(name) and other info are placed into the `char from[256]` in initreqprep. If the name was too long, the addr-spec and params wouldn't fit. Code is moved around so the addr-spec with params is placed there first, and then fitting in as much of the display-name as possible. ASTERISK-25396 #close Change-Id: I33632baf024f01b6a00f8c7f35c91e5f68c40260	10 years ago
Richard Mudgett	4cc59533b9	CHAOS: res_pjsip_diversion avoid crash if allocation fails ... Validate ast_malloc buffer returned before using it in set_redirecting_value(). ASTERISK-25323 Change-Id: I15d2ed7cb0546818264c0bf251aa40adeae83253	10 years ago
Kevin Harwell	4fb95bbc4e	app_queue: AgentComplete event has wrong reason ... When a queued caller transfers an agent to another extension sometimes the raised AgentComplete event has a reason of "caller" and sometimes "transfer". Since a transfer has taken place this should always be transfer. This occurs because sometimes the stasis hangup event arrives before the transfer event thus writing a different reason out. With this patch, when a hangup event is received during a transfer it will check to see if the channel that is hanging up is part of a transfer. If so it will return and let the subsequently received transfer event handler take care of the cleanup. ASTERISK-25399 #close Change-Id: Ic63c49bd9a5ed463ea7a032fd2ea3d63bc81a50d	10 years ago
Scott Griepentrog	fb6b5c684b	PJSIP: avoid crash when getting rtp peer ... Although unlikely, if the tech private is returned as a NULL, chan_pjsip_get_rtp_peer() would crash. ASTERISK-25323 Change-Id: Ie231369bfa7da926fb2b9fdaac228261a3152e6a	10 years ago
Joshua Colp	a665b31281	Merge "res_pjsip_pubsub: Eliminate race during initial NOTIFY." into 13	10 years ago
Kevin Harwell	6409e7b11a	app_queue: Crash when transferring ... During some transfer scenarios involving queues Asterisk would sometimes crash when trying to obtain a channel snapshot (could happen on caller or member channels). This occurred because the underlying channel had already disappeared when trying to obtain the latest snapshot. This patch adds a reference to both the member and caller channels that extends to the lifetime of the queue'd call, thus making sure the channels will always exist when retrieving the latest snapshots. ASTERISK-25185 #close Reported by: Etienne Lessard Change-Id: Ic397fa68fb4ff35fbc378e745da9246a7b552128	10 years ago
Mark Michelson	fe5077b1f8	res_pjsip_pubsub: Eliminate race during initial NOTIFY. ... There is a slim chance of a race condition occurring where two threads can both attempt to manipulate the same area. Thread A can be handling an incoming initial SUBSCRIBE request. Thread A lets the specific subscription handler know that the subscription has been established. At this point, Thread B may detect a state change on the subscribed resource and queue up a notification task on Thread C, the subscription serializer thread. Now Thread A attempts to generate the initial NOTIFY request to send to the subscriber at the same time that Thread C attempts to generate a state change NOTIFY request to send to the subscriber. The result is that Threads A and C can step on the same memory area, resulting in a crash. The crash has been observed as happening when attempting to allocate more space to hold the body for the NOTIFY. The solution presented here is to queue the subscription establishment and initial NOTIFY generation onto the subscription serializer thread (Thread C in the above scenario). This way, there is no way that a state change notification can occur before the initial NOTIFY is sent, and if there is a quick succession of NOTIFYs, we can guarantee that the two NOTIFY requests will be sent in succession. Change-Id: I5a89a77b5f2717928c54d6efb9955e5f6f5cf815	10 years ago
Alexander Traud	b88c54fa4b	translate: Fix transcoding while different in frame size. ... When Asterisk translates between codecs, each with a different frame size (for example between iLBC 30 and Speex-WB), too large frames were created by ast_trans_frameout. Now, ast_trans_frameout is called with the correct frame length, creating several frames when necessary. Affects all transcoding modules which used ast_trans_frameout: GSM, iLBC, LPC10, and Speex. ASTERISK-25353 #close Change-Id: I2e229569d73191d66a4e43fef35432db24000212	10 years ago
Mark Michelson	5c713fdf18	scheduler: Use queue for allocating sched IDs. ... It has been observed that on long-running busy systems, a scheduler context can eventually hit INT_MAX for its assigned IDs and end up overflowing into a very low negative number. When this occurs, this can result in odd behaviors, because a negative return is interpreted by callers as being a failure. However, the item actually was successfully scheduled. The result may be that a freed item remains in the scheduler, resulting in a crash at some point in the future. The scheduler can overflow because every time that an item is added to the scheduler, a counter is bumped and that counter's current value is assigned as the new item's ID. This patch introduces a new method for assigning scheduler IDs. Instead of assigning from a counter, a queue of available IDs is maintained. When assigning a new ID, an ID is pulled from the queue. When a scheduler item is released, its ID is pushed back onto the queue. This way, IDs may be reused when they become available, and the growth of ID numbers is directly related to concurrent activity within a scheduler context rather than the uptime of the system. Change-Id: I532708eef8f669d823457d7fefdad9a6078b99b2	10 years ago
Matt Jordan	4fd0af298e	Merge "res/res_pjsip_nat: Ignore REGISTER requests when looking for a Record-Route" into 13	10 years ago
Rodrigo Ramírez Norambuena	865377fc38	chan_sip.c: Validation on module reload ... Change validation on reload module because now used the cli function for reload. The sip_reload() function never fail and ever return NULL for this reason on reload() now use the call the sip_reload() and return AST_MODULE_LOAD_SUCCESS. This problem is dectected on reload by PUT method on ARI, getting always 404 http code when the module is reloaded. ASTERISK-25325 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: I41215877fb2cfc589e0d4d464000cf6825f4d7fb	10 years ago
Richard Mudgett	e75aff53e6	res_pjsip_pubsub.c: Mark ast_sip_create_subscription() as not used. ... Change-Id: I2b8db18eac36c01a5c7eb9467699124e203fd093	10 years ago
Richard Mudgett	4d91d01df1	res_pjsip_pubsub.c: Add some notification comments. ... Change-Id: Ie62ff1f4b7adc1a12fa0303f53926af249b25e20	10 years ago
Richard Mudgett	f36a9d1221	res_pjsip_pubsub.c: Set dlg_status code instead of sending SIP response. ... We should not try to send a SIP response message because we may be restoring a persistent subscription where we are not responding to a SIP request. Change-Id: Id89167ef90320c5563f37e632db0dda6cb9e7dec	10 years ago
Richard Mudgett	94582f8fab	res_pjsip_pubsub.c: Fix off-nominal memory leak. ... Fix off-nominal visited vector leak in build_resource_tree(). Change-Id: If0399c7941c9c0b1038bcfb7b9a371760977831c	10 years ago
Richard Mudgett	8b3ed52239	res_pjsip_pubsub.c: Fix one byte buffer overrun error. ... ast_sip_pubsub_register_body_generator() did not account for the null terminator set by sprintf() in the allocated output buffer. Change-Id: I388688a132e479bca6ad1c19275eae0070969ae2	10 years ago
Richard Mudgett	4329bd1e4c	res_pjsip_pubsub.c: Use ast_alloca() instead of alloca(). ... Change-Id: Ia396096b4fedc2874649ca11137612c3f55e83e3	10 years ago
Richard Mudgett	a456a20ecf	res_pjsip_pubsub.c: Add missing error return in load_module(). ... Change-Id: I15debd0f717f16ee2f78e7f56151c3b3b97b72fc	10 years ago
Richard Mudgett	f58f4c6e27	res_pjsip/location.c: Use the builtin ao2_callback() match function instead. ... Change-Id: I364906d6d2bad3472929986704a0286b9a2cbe3f	10 years ago
Mark Michelson	9d1f176e29	res_pjsip: Copy default_from_user to avoid crash. ... The default_from_user retrieval function was pulling the default_from_user from the global configuration struct in an unsafe way. If using a database as a backend configuration store, the global configuration struct is short-lived, so grabbing a pointer from it results in referencing freed memory. The fix here is to copy the default_from_user value out of the global configuration struct. Thanks go to John Hardin for discovering this problem and proposing the patch on which this fix is based. ASTERISK-25390 #close Reported by Mark Michelson Change-Id: I6b96067a495c1259da768f4012d44e03e7c6148c	10 years ago
Matt Jordan	1dd0e220bf	res/res_pjsip_nat: Ignore REGISTER requests when looking for a Record-Route ... We will only rewrite the Contact header if there is no Record-Route header in the received request. If a malfunctioning proxy places a Record-Route header into a REGISTER request, we will decide that we shouldn't update the IP/port in the Contact header, and we will end up storing a contact with an AoR that contains the NAT'd IP address. While it is nice to have the proxy *not* send a Record-Route in a REGISTER request, it's also a good idea to not process the header in a non-dialog message. This patch updates the code to explicitly ignore the Record-Route header in REGISTER requests. ASTERISK-25387 #close Change-Id: I4bd3bcccc4003d460cc354d986b0dea2e433ef3f	10 years ago
Joshua Colp	16fa1cbb6c	Merge "ParkAndAnnounce: Add variable inheritance" into 13	10 years ago