4977 Commits (ced0a2d71b690f24026392bcfbbe4c36eb8d4dff)

Author	SHA1	Message	Date
Joshua Colp	771757e338	Merge "translate: Fix transcoding while different in frame size." into 13	11 years ago
Joshua Colp	41f856e5a2	Merge "logger: Prevent duplicate dynamic channels from being added." into 13	11 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	11 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	11 years ago
Matt Jordan	0d95610ae2	Merge "ARI: Add events for Contact and Peer Status changes" into 13	11 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	11 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	11 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	11 years ago
Joshua Colp	a45abbf3ce	Merge "pbx: Update device and presence state when changing a hint extension." into 13	11 years ago
Matt Jordan	960f00939a	Merge "main/config_options: Check for existance of internal object before derefing" into 13	11 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	11 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	11 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	11 years ago
Matt Jordan	4eedd9ef9d	main/config_options: Check for existance of internal object before derefing ... Asterisk can load and register an object type while still having an invalid sorcery mapping. This can cause an issue when a creation call is invoked. For example, mis-configuring PJSIP's endpoint identifier by IP address mapping in sorcery.conf will cause the sorcery mechanism to be invalidated; however, a subsequent ARI invocation to create the object will cause a crash, as the internal type may not be registered as sorcery expects. Merely checking for a NULL pointer here solves the issue. Change-Id: I54079fb94a1440992f4735a9a1bbf1abb1c601ac	11 years ago
Guido Falsi	4ed9c9a280	Core/General: Add #ifdef needed on FreeBSD. ... pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED on FreeBSD too. ASTERISK-25310 #close Reported by: Guido Falsi Change-Id: Iae6befac9028b5b9795f86986a4a08a1ae6ab7c4	11 years ago
Scott Griepentrog	6002472a62	endpoint snapshot: avoid second cleanup on alloc failure ... In ast_endpoint_snapshot_create(), a failure to init the string fields results in two attempts to ao2_cleanup the same pointer. Removed RAII_VAR to eliminate problem. ASTERISK-25375 #close Reported by: Scott Griepentrog Change-Id: If4d9dfb1bbe3836b623642ec690b6d49b25e8979	11 years ago
Joshua Colp	cc1363209e	pbx: Fix crash when issuing "core show hints" with long pattern match. ... When issuing the "core show hints" CLI command a combination of both the hint extension and context is created. This uses a fixed size buffer expecting that the extension will not exceed maximum extension length. When the extension is actually a pattern match this constraint does not hold true, and the extension may exceed the maximum extension length. In this case extra characters are written past the end of the fixed size buffer. This change makes it so the construction of the combined hint extension and context can not exceed the size of the buffer. ASTERISK-25367 #close Change-Id: Idfa1b95d0d4dc38e675be7c1de8900b3f981f499	11 years ago
Joshua Colp	4e3b3f25f6	Merge "Fix deadlock on presence state changes." into 13	11 years ago
Mark Michelson	03fe79f29e	Fix deadlock on presence state changes. ... A deadlock was observed where three threads were competing for different locks: * One thread held the hints lock and was attempting to lock a specific hint. * One thread was holding the specific hint's lock and was attempting to lock the contexts lock * One thread was holding the contexts lock and attempting to lock the hints lock. Clearly the second thread was doing the wrong thing here. The fix for this is to make sure that the hint's lock is not held on presence state changes. Something similar is already done (and commented about) for device state changes. ASTERISK-25362 #close Reported by Mark Michelson Change-Id: I15ec2416b92978a4c0c08273b2d46cb21aff97e2	11 years ago
Joshua Colp	a676ba2aad	taskprocessor: Fix race condition between unreferencing and finding. ... When unreferencing a taskprocessor its reference count is checked to determine if it should be unlinked from the taskprocessors container and its listener shut down. In between the time when the reference count is checked and unlinking it is possible for another thread to jump in, find it, and get a reference to it. If the thread then uses the taskprocessor it may find that it is not in the state it expects. This change locks the taskprocessors container during almost the entire unreference operation to ensure that any other thread which may attempt to find the taskprocessor has to wait. ASTERISK-25295 Change-Id: Icb842db82fe1cf238da55df92e95938a4419377c	11 years ago
Joshua Colp	85e1cb51b2	sched: ast_sched_del may return prematurely due to spurious wakeup ... When deleting a scheduled item if the item in question is currently executing the ast_sched_del function waits until it has completed. This is accomplished using ast_cond_wait. Unfortunately the ast_cond_wait function can suffer from spurious wakeups so the predicate needs to be checked after it returns to make sure it has really woken up as a result of being signaled. This change adds a loop around the ast_cond_wait to make sure that it only exits when the executing task has really completed. ASTERISK-25355 #close Change-Id: I51198270eb0b637c956c61aa409f46283432be61	11 years ago
Joshua Colp	6c2dab1e88	bridge: Kick channel from bridge if hung up during action. ... When executing an action in a bridge it is possible for the channel to be hung up without the bridge becoming aware of it. This is most easily reproducible by hanging up when the bridge is streaming DTMF due to a feature timeout. This change makes it so after action execution the channel is checked to determine if it has been hung up and if it has it is kicked from the bridge. ASTERISK-25341 #close Change-Id: I6dd8b0c3f5888da1c57afed9e8a802ae0a053062	11 years ago
Scott Griepentrog	ab373f2cef	CHAOS: prevent sorcery object with null id ... When allocating a sorcery object, fail if the id value was not allocated. ASTERISK-25323 Reported by: Scott Griepentrog Change-Id: I152133fb7545a4efcf7a0080ada77332d038669e	11 years ago
Richard Mudgett	cea5dc7b8a	audiohook.c: Simplify variable usage in audiohook_read_frame_both(). ... Change-Id: I58bed58631a94295b267991c5b61a3a93c167f0c	11 years ago
Richard Mudgett	b3a56bee83	audiohook.c: Fix MixMonitor crash when using the r() or t() options. ... The built frame format in audiohook_read_frame_both() is now set to a signed linear format before the rx and tx frames are duplicated instead of only for the mixed audio frame duplication. ASTERISK-25322 #close Reported by Sean Pimental Change-Id: I86f85b5c48c49e4e2d3b770797b9d484250a1538	11 years ago
Matt Jordan	a0f451c35e	main/format: Add an API call for retrieving format attributes ... Some codecs that may be a third party library to Asterisk need to have knowledge of the format attributes that were negotiated. Unfortunately, when the great format migration of Asterisk 13 occurred, that ability was lost. This patch adds an API call, ast_format_attribute_get, to the core format API, along with updates to the unit test to check the new API call. A new callback is also now available for format attribute modules, such that they can provide the format attribute values they manage. Note that the API returns a void *. This is done as the format attribute modules themselves may store format attributes in any particular manner they like. Care should be taken by consumers of the API to check the return value before casting and dereferencing. Consumers will obviously need to have a priori knowledge of the type of the format attribute as well. Change-Id: Ieec76883dfb46ecd7aff3dc81a52c81f4dc1b9e3	11 years ago
Matt Jordan	c3bd7fb835	Merge "rtp_engine.c: Fix performance issue with several channel drivers that use RTP." into 13	11 years ago
Scott Emidy	df9ce36366	ARI: Retrieve existing log channels ... An http request can be sent to get the existing Asterisk logs. The command "curl -v -u user:pass -X GET 'http://localhost:8088 /ari/asterisk/logging'" can be run in the terminal to access the newly implemented functionality. * Retrieve all existing log channels ASTERISK-25252 Change-Id: I7bb08b93e3b938c991f3f56cc5d188654768a808	11 years ago
Scott Emidy	e9f1bc08cb	ARI: Creating log channels ... An http request can be sent to create a log channel in Asterisk. The command "curl -v -u user:pass -X POST 'http://localhost:088/ari/asterisk/logging/mylog? configuration=notice,warning'" can be run in the terminal to access the newly implemented functionality for ARI. * Ability to create log channels using ARI ASTERISK-25252 Change-Id: I9a20e5c75716dfbb6b62fd3474faf55be20bd782	11 years ago
Joshua Colp	cf27200391	Merge "ARI: Deleting log channels" into 13	11 years ago
Scott Emidy	78364132ce	ARI: Deleting log channels ... An http request can be sent to delete a log channel in Asterisk. The command "curl -v -u user:pass -X DELETE 'http://localhost:8088 /ari/asterisk/logging/mylog'" can be run in the terminal to access the newly implemented functionally for ARI. * Able to delete log channels using ARI ASTERISK-25252 Change-Id: Id6eeb54ebcc511595f0418d586ff55914bc3aae6	11 years ago
Joshua Colp	ba40b07ddc	Merge topic 'misc_rtp_tweaks' into 13 ... * changes: rtp_engine.c: Must protect mime_types_len with mime_types_lock. res_pjsip_sdp_rtp.c: Fixup some whitespace.	11 years ago
Joshua Colp	20ee33e22e	Merge topic 'misc_rtp_tweaks' into 13 ... * changes: rtp_engine.h: No sense allowing payload types larger than RFC allows. rtp_engine.c: Minor tweaks. rtp_engine.h: Misc comment fixes. chan_sip.c: Tweak glue->update_peer() parameter nil value.	11 years ago
Mark Michelson	e28fbebc57	Merge "ARI: Rotate log channels." into 13	11 years ago
Benjamin Ford	1ae762634c	ARI: Rotate log channels. ... An http request can be sent to rotate a specified log channel. If the channel does not exist, an error response will be returned. The command "curl -v -u user:pass -X PUT 'http://localhost:8088 /ari/asterisk/logging/logChannelName/rotate'" can be run in the terminal to access this new functionality. * Added the ability to rotate log files through ARI ASTERISK-25252 Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01	11 years ago
Richard Mudgett	aeeb170fc4	rtp_engine.c: Fix performance issue with several channel drivers that use RTP. ... ast_rtp_codecs_get_payload() gets called once or twice for every received RTP frame so it would be nice to not allocate an ao2 object to then have it destroyed shortly thereafter. The ao2 object gets allocated only if the payload type is not set by the channel driver as a negotiated value. The issue affects chan_skinny, chan_unistim, chan_rtp, and chan_ooh323. * Made static_RTP_PT[] an array of ao2 objects that ast_rtp_codecs_get_payload() can return instead of an array of structs that must be copied into a created ao2 object. ASTERISK-25296 #close Reported by: Richard Mudgett Change-Id: Icb6de5cd90bfae07d44403a1352963db9109dac0	11 years ago
Richard Mudgett	1519eb44a7	rtp_engine.c: Must protect mime_types_len with mime_types_lock. ... Change-Id: I44220dd369cc151ebf5281d5119d84bb9e54d54e	11 years ago
Richard Mudgett	89b21fd9a3	rtp_engine.h: No sense allowing payload types larger than RFC allows. ... * Tweaked add_static_payload() to not use magic numbers. Change-Id: I1719ff0f6d3ce537a91572501eae5bcd912a420b	11 years ago
Richard Mudgett	7427c7f13b	rtp_engine.c: Minor tweaks. ... * Fix off nominial ref leak of new_type in ast_rtp_codecs_payloads_set_m_type(). * No need to lock static_RTP_PT_lock in ast_rtp_codecs_payloads_set_m_type() and ast_rtp_codecs_payloads_set_rtpmap_type_rate() before the payload type parameter sanity check. * No need to create ast_rtp_payload_type ao2 objects with a lock since the lock is not used. Change-Id: I64dd1bb4dfabdc7e981e3f61448beac9bb7504d4	11 years ago
Mark Michelson	10ba72a927	Add a test event for inband ringing. ... This event is necessary for the bridge_wait_e_options test to be able to confirm that ringing is being played on the local channel that runs the BridgeWait() application with the e(r) option. ASTERISK-25292 #close Reported by Kevin Harwell Change-Id: Ifd3d3d2bebc73344d4b5310d0d55c7675359d72e	11 years ago
Mark Michelson	c9099d06cc	Merge "holding_bridge: ensure moh participants get frames" into 13	11 years ago
Jonathan Rose	8458b8d441	holding_bridge: ensure moh participants get frames ... Currently, if a blank musiconhold.conf is used, musiconhold will fail to start for a channel going into a holding bridge with an anticipation of getting music on hold. That being the case, no frames will be written to the channel and that can pose a problem for blind transfers in PJSIP which may rely on frames being written to get past the REFER framehook. This patch makes holding bridges start a silence generator if starting music on hold fails and makes it so that if no music on hold functions are installed that the ast_moh_start function will report a failure so that consumers of that function will be able to respond appropriately. ASTERISK-25271 #close Change-Id: I06f066728604943cba0bb0b39fa7cf658a21cd99	11 years ago
Joshua Colp	2749721791	pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options. ... This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold' endpoint options. These allow the channel to be hung up if RTP is not received from the remote endpoint for a specified number of seconds. ASTERISK-25259 #close Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9	11 years ago
Mark Michelson	f635520527	Local channels: Alternate solution to ringback problem. ... Commit 54b25c80c8 solved an issue where a specific scenario involving local channels and a native local RTP bridge could result in ringback still being heard on a calling channel even after the call is bridged. That commit caused many tests in the testsuite to fail with alarming consequences, such as not sending DialBegin and DialEnd events, and giving incorrect hangup causes during calls. This commit reverts the previous commit and implements and alternate solution. This new solution involves only passing AST_CONTROL_RINGING frames across local channels if the local channel is in AST_STATE_RING. Otherwise, the frame does not traverse the local channels. By doing this, we can ensure that a playtones generator does not get started on the calling channel but rather is started on the local channel on which the ringing frame was initially indicated. ASTERISK-25250 #close Reported by Etienne Lessard Change-Id: I3bc87a18a38eb2b68064f732d098edceb5c19f39	11 years ago
Matt Jordan	4d8f47f4bf	Merge "audiohook: Use manipulated frame instead of dropping it." into 13	11 years ago
Joshua Colp	ff83c115c7	Merge "Local channels: Do not block control -1 payloads." into 13	11 years ago
Joshua Colp	f509730cb9	audiohook: Use manipulated frame instead of dropping it. ... Previous changes to sample rate support in audiohooks accidentally removed code responsible for allowing the manipulate audiohooks to work. Without this code the manipulated frame would be dropped and not used. This change restores it. ASTERISK-25253 #close Change-Id: I3ff50664cd82faac8941f976fcdcb3918a50fe13	11 years ago
Mark Michelson	54b25c80c8	Local channels: Do not block control -1 payloads. ... Control frames with a -1 payload are used as a special signal to stop playtones generators on channels. This indication is sent both by app_dial as well as by ast_answer() when a call is answered in case any tones were being generated on a calling channel. This control frame type was made to stop traversing local channel pairs as an optimization, because it was thought that it was unnecessary to send these indications, and allowing such unnecessary control frames to traverse the local channels would cause the local channels to optimize away less quickly. As it turns out, through some special magic dialplan code, it is possible to have a tones being played on a non-local channel, and it is important for the local channel to convey that the tones should be stopped. The result of having tones continue to be played on the non-local channel is that the tones play even once the channel has been bridged. By not blocking the -1 control frame type, we can ensure that this situation does not happen. ASTERISK-25250 #close Reported by Etienne Lessard Change-Id: I0bcaac3d70b619afdbd0ca8a8dd708f33fd2f815	11 years ago
Joshua Colp	f1493f900e	audiohook: Read the correct number of samples based on audiohook format. ... Due to changes in audiohooks to support different sample rates the underlying storage of samples is in the format of the audiohook itself and not of the format being requested. This means that if a channel is using G722 the samples stored will be at 16kHz. If something subsequently reads from the audiohook at a format which is not the same sample rate as the audiohook the number of samples needs to be adjusted. Given the following example: 1. Channel writing into audiohook at 16kHz (as it is using G722). 2. Chanspy reading from audiohook at 8kHz. The original code would read 160 samples from the audiohook for each 20ms of audio. This is incorrect. Since the audio in the audiohook is at 16kHz the actual number needing to be read is 320. Failure to read this much would cause the audiohook to reset itself constantly as the buffer became full. This change adjusts the requested number of samples by determining the duration of audio requested and then calculating how many samples that would be in the audiohook format. ASTERISK-25247 #close Change-Id: Ia91ce516121882387a315fd8ee116b118b90653d	11 years ago
Mark Michelson	d9094ddd73	res_pjsip: Add rtp_keepalive endpoint option. ... This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the chan_sip option, this specifies an interval, in seconds, at which we will send RTP comfort noise frames. This can be useful for keeping RTP sessions alive as well as keeping NAT associations alive during lulls. ASTERISK-25242 #close Reported by Mark Michelson Change-Id: I06660ba672c0a343814af4cec838e6025cafd54b	11 years ago