6464 Commits (3e0a2485c4110d5bcd765bd607bbdafbc1af1e2a)

Author	SHA1	Message	Date
Friendly Automation	2770830495	Merge "main/file.c: Limit media cache usage to remote files." into 16	7 years ago
Friendly Automation	a3edac10a6	Merge "confbridge: Add support for specifying maximum sample rate." into 16	7 years ago
Joshua C. Colp	5622df0a94	confbridge: Add support for specifying maximum sample rate. ... ConfBridge has the ability to move between different sample rates for mixing the conference bridge. Up until now there has only been the ability to set the conference bridge to mix at a specific sample rate, or to let it move between sample rates as necessary. This change adds the ability to configure a conference bridge with a maximum sample rate so it can move between sample rates but only up to the configured maximum. ASTERISK-28658 Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee	7 years ago
Jaco Kroon	77941efad9	ACL: ast_apply_acl_nolog - identical to ast_apply_acl but without logging. ... Due to use in res_rtp_asterisk there is a need to be able to apply an ACL without logging any invalid/denies. It's probably sensible to at least validate the ACL once directly after load and report invalid ACLs. Change-Id: I256169229d945ca7c1bbf228fc492d91df345843 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	7 years ago
George Joseph	7f2bbff48c	Merge "channel.c: Resolve issue with receiving SIP INFO packets for DTMF" into 16	7 years ago
George Joseph	43d4c0e3c9	channel.c: Resolve issue with receiving SIP INFO packets for DTMF ... The problem is essentially the same as in ASTERISK~28245. Besides the direct media scenario we have an additional scenario where a special client is involved. This device mutes audio by default in transmit direction (no rtp frames) and activates audio only by a foot switch. In this situation dtmf input (pin for conferences, transfer features codes , etc) using SIP INFO mode is not understood properly especially when SIP INFO messages are sent quickly. This patch ensures that SIP INFO frames are properly queued and processed in the above scenario. The patch also corrects situations where successive dtmf events are received quicker than the signalled event duration (plus minimum gap/pause) allows, i.e. DTMF events have to be buffered in the ast channel read queue and emulation has to be processed asynchronously at slower speed. Reported by: Thomas Arimont patches: trigger_dtmf_emulation.patch submitted by Thomas Arimont (license 5525) Change-Id: I309bf61dd065c9978c8e48f5b9a936ab47de64c2	7 years ago
Sean Bright	bd3cb1b300	media_cache.c: Various CLI improvements ... * Use ast_cli_completion_add() to improve performance when large number of cached items are present. * Only complete one URI for commands that only accept a single URI. * Change command documentation to wrap at 80 characters to improve readability. Change-Id: Iedb0a2c3541e49561bc231dca2dcc0ebd8612902	7 years ago
Kevin Reeves	e013f502b1	main/file.c: Limit media cache usage to remote files. ... When testing for the existance of a file, the media cache is searched even if the file has no chance of being in it. This can cause performance issues as the media cache size increases. As a result, calls to applications like Read and Playback using local files must scan through the media cache before playing. Under load and with a large cache, this can delay the playback of those files. This patch updates the function that checks for the existance of a file to only consult the media cache database if the requested file is a remote path. It introduces a new is_remote_path() function in main/file.c. ASTERISK-28625 #close Reported-by: kevin@phoneburner.com Change-Id: If91137493732d9034dafa381c081c69274a7dcc9	7 years ago
George Joseph	7574be5110	manager.c: Prevent the Originate action from running the Originate app ... If an AMI user without the "system" authorization calls the Originate AMI command with the Originate application, the second Originate could run the "System" command. Action: Originate Channel: Local/1111 Application: Originate Data: Local/2222,app,System,touch /tmp/owned If the "system" authorization isn't set, we now block the Originate app as well as the System, Exec, etc. apps. ASTERISK-28580 Reported by: Eliel Sardañons Change-Id: Ic4c9dedc34c426f03c8c14fce334a71386d8a5fa	7 years ago
Friendly Automation	b7c0711c5c	Merge "serializer: set high/low alert levels on whole pool" into 16	7 years ago
Kevin Harwell	30c0af7257	various files - fix some alerts raised by lgtm code analysis ... This patch fixes several issues reported by the lgtm code analysis tool: https://lgtm.com/projects/g/asterisk/asterisk Not all reported issues were addressed in this patch. This patch mostly fixes confirmed reported errors, potential problematic code points, and a few other "low hanging" warnings or recommendations found in core supported modules. These include, but are not limited to the following: * innapropriate stack allocation in loops * buffer overflows * variable declaration "hiding" another variable declaration * comparisons results that are always the same * ambiguously signed bit-field members * missing header guards Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25	7 years ago
Alexei Gradinari	6be18dfb72	serializer: set high/low alert levels on whole pool ... The current code sets alert levels starting from index 1. Need to set on whole pool starting from index 0. Change-Id: I5decbb43160954fb9a512f04302637fc666b6f5d	7 years ago
Friendly Automation	e7e03e0e2e	Merge "pbx: deadlock when outgoing dialed channel hangs up too quickly" into 16	7 years ago
Kevin Harwell	a66848c92f	pbx: deadlock when outgoing dialed channel hangs up too quickly ... Here's the basic scenario that occurred when executing an AMI fast originate while at the same time something else locks the channels container, and also wants a lock on the dialed channel: 1. pbx_outgoing_attempt obtains a lock on a dialed channel 2. concurrently another thread obtains a lock on the channels container, and subsequently requests a lock on the dialed channel. It waits on #1. For instance, "core show channel <dialed channel" 3. the outgoing call does not fail, but ends before the pbx_outgoing_attempt function exits 4. pbx_outgoing_attempt function exits, the outgoing structure destructs, and attempts to hang up the dialed channel 5. hang up tries to obtain the channels container lock, but can't due to #2. 6. Asterisk is deadlocked. The solution was to allow the pbx_outgoing_exec function to "steal" ownership of the dialed channel, and handle hanging it up. The channel now is either hung up prior to it being potentially locked by the initiating thread, or if locked the hang up takes place in a different thread, thus alleviating the deadlock. ASTERISK-28561 patches: iliketrains.diff submitted by Joshua Colp (license 5000) Change-Id: I51b42b92dde8f2215b69bb509e28667ee3a3853a	7 years ago
Kevin Harwell	afc10c25ac	serializer: move/add asterisk serializer pool functionality ... Serializer pools have previously existed in Asterisk. However, for the most part the code has been duplicated across modules. This patch abstracts the code into an 'ast_serializer_pool' object. As well the code is now centralized in serializer.c/h. In addition serializer pools can now optionally be monitored by a shutdown group. This will prevent the pool from being destroyed until all serializers have completed. Change-Id: Ib1e906144b90ffd4d5ed9826f0b719ca9c6d2971	7 years ago
Joshua Colp	ce1e0714ba	stasis: Pass bumped topic_all reference to proxy_dtor. ... This avoids use of the global variable and ensures topic_all remains active until all topics are freed. ASTERISK-28553 patches: ASTERISK-28553.patch by coreyfarrell (license 5909) Change-Id: I9a8cd8977f3c3a6aa00783f8336d2cfb9c2820f1	7 years ago
George Joseph	ac331bff34	Merge "pbx: Prevent Realtime switch crash on invalid priority" into 16	7 years ago
Sean Bright	0514559005	pbx: Prevent Realtime switch crash on invalid priority ... pbx_extension_helper takes two 'context' arguments. One (con) is a pointer directly to a 'struct ast_context' and the other (context) is the name of the context. In all cases, one of these arguments is NULL and the other is non-NULL. Functions that are ultimately called by pbx_extension_helper expect that 'context' will be non-NULL, so we set it unconditionally on entry into this function. ASTERISK-28534 #close Change-Id: Ifbbc5e71440afd80efd441f7a9d72e8b10b6f47d	7 years ago
Ben Ford	827dd754b2	taskprocessor.c: Added "like" support to 'core show taskprocessors' ... Added "like" support for 'core show taskprocessors'. Now you can specify a specific set of taskprocessors (or just one) by adding the keyword "like" to the above command, followed by your search criteria. Change-Id: I021e740201e9ba487204b5451e46feb0e3222464	7 years ago
Friendly Automation	0b34551af0	Merge "core: Fix ABI mismatch of ao2_global_obj." into 16	7 years ago
George Joseph	cce4dd2e71	Merge "taskprocessor.c: Add CLI commands to reset taskprocessor stats." into 16	7 years ago
George Joseph	d799217867	Merge "core: Add AO2_ALLOC_OPT_NO_REF_DEBUG option." into 16	7 years ago
Corey Farrell	cd51f5b876	core: Fix ABI mismatch of ao2_global_obj. ... astobj2.c declares DEBUG_THREADS_LOOSE_ABI to avoid overhead of debug threads tracking information in the internal structures of astobj2. Unfortunately this means that ao2_global_obj contains the statically allocated debug threads tracking fields which are used by initialization and cleanup but main/astobj2.c believed those fields and associated space did not exist. Change-Id: Icef41ad97d88a8c1d1515e034ec8133cab3b1527	7 years ago
Ben Ford	5ea667e03a	taskprocessor.c: Add CLI commands to reset taskprocessor stats. ... Added two new CLI commands to reset stats for taskprocessors. You can reset stats for a single, specific taskprocessor ('core reset taskprocessor <taskprocessor>'), or you can reset all taskprocessors ('core reset taskprocessors'). These commands will reset the counter for the number of tasks processed as well as the max queue size. Change-Id: Iaf17fc4ae29396ab0c6ac92408fc7bdc2f12362d	7 years ago
George Joseph	b41e1e8928	Merge "astmm.c: Display backtrace with memory show allocations" into 16	7 years ago
Kevin Harwell	258f9e6ecc	Merge "res_pjsip_mwi: Remove inappropriate topic unreference." into 16	7 years ago
Corey Farrell	fec6e1bd87	core: Add AO2_ALLOC_OPT_NO_REF_DEBUG option. ... Previous to this patch passing a NULL tag to ao2_alloc or ao2_ref based functions would result in the reference not being logged under REF_DEBUG. This could sometimes cause inaccurate logging if NULL was accidentally passed to a reference action. Now reference logging is only disabled by option passed to the allocation method. Change-Id: I3c17d867d901d53f9fcd512bef4d52e342637b54	7 years ago
George Joseph	9c5a8066a6	astmm.c: Display backtrace with memory show allocations ... You can currently capture backtraces of memory allocations but they only get displayed when you stop asterisk and the atexit hooks are enabled. Now, if memory backtrace is on and you issue a "memory show allocations" CLI command for a specific file, then a backtrace will show for each allocation that occurred after you turned "memory backtrace on". The backtrace display is shown only when a specific file's allocations are displayed to prevent a massive CLI dump of every file's allocations. Change-Id: Ic657afc1fc6ec7205e16eb36a97a611d235a2b4f	7 years ago
Corey Farrell	5b8c1ed0d3	stasis: refcounter.py can incorrectly report skewed objects. ... It is possible for topic->name to be NULL, this causes the allocation reference to not be logged. Use the name variable instead which has been verified to be a non-empty. Change-Id: I3d0031d03c8356e4808f00cdf2d5428712575883	7 years ago
Corey Farrell	76d4a42ae1	res_pjsip_mwi: Remove inappropriate topic unreference. ... ast_mwi_topic() returns a borrowed reference which should not be unreferenced, doing so leads to a FRACK. This was hidden by the fact that stasis_cache.c leaked the result of cache_remove in caching_topic_exec. Change-Id: I51101bf7d07b8dc8ce8fc46b6cb31fbbd213fbc7	7 years ago
Friendly Automation	a687c7919d	Merge "func_jitterbuffer: Add audio/video sync support." into 16	7 years ago
Joshua Colp	120f2fb5f9	Merge "core: Add H.265/HEVC passthrough support" into 16	7 years ago
Joshua Colp	6647be69ac	func_jitterbuffer: Add audio/video sync support. ... This change adds support to the JITTERBUFFER dialplan function for audio and video synchronization. When enabled the RTCP SR report is used to produce an NTP timestamp for both the audio and video streams. Using this information the video frames are queued until their NTP timestamp is equal to or behind the NTP timestamp of the audio. The audio jitterbuffer acts as the leader deciding when to shrink/grow the jitterbuffer when adaptive is in use. For both adaptive and fixed the video buffer follows the size of the audio jitterbuffer. ASTERISK-28533 Change-Id: I3fd75160426465e6d46bb2e198c07b9d314a4492	7 years ago
George Joseph	913c8b48b7	Merge "channels: Allow updating variable value" into 16	7 years ago
George Joseph	c2dbba39a6	Merge "res_rtp: Add unit tests for RTCP stats." into 16	7 years ago
Sean Bright	518b6bfb5c	channels: Allow updating variable value ... When modifying an already defined variable in some channel drivers they add a new variable with the same name to the list, but that value is never used, only the first one found. Introduce ast_variable_list_replace() and use it where appropriate. ASTERISK-23756 #close Patches: setvar-multiplie.patch submitted by Michael Goryainov Change-Id: Ie1897a96c82b8945e752733612ee963686f32839	7 years ago
Florian Floimair	f85631cf82	core: Add H.265/HEVC passthrough support ... This change adds H.265/HEVC as a known codec and creates a cached "h265" media format for use. Note that RFC 7798 section 7.2 also describes additional SDP parameters. Handling of these is not yet supported. ASTERISK-28512 Change-Id: I26d262cc4110b4f7e99348a3ddc53bad0d2cd1f2	7 years ago
Ben Ford	922d3e02df	res_rtp: Add unit tests for RTCP stats. ... Added unit tests for RTCP video stats. These tests include NACK, REMB, FIR/FUR/PLI, SR/RR/SDES, and packet loss statistics. The REMB and FIR tests are currently disabled due to a bug. We expect to receive a compound packet, but the code sends this out as a single packet, which the browser accepts, but makes Asterisk upset. While writing these tests, I noticed an issue with NACK as well. Where it is handling a received NACK request, it was reading in only the first 8 bits of following packets that were also lost. This has been changed to the correct value of 16 bits. Also made a minor fix to the data buffer unit test. Change-Id: I56107c7411003a247589bbb6086d25c54719901b	7 years ago
Joshua Colp	2691ee7e10	AST-2019-005 - translate: Don't assume all frames will have a src. ... This change removes the assumption that a frame will always have a src set on it. This assumption is incorrect. Given a scenario where an RTP packet is received with no payload the resulting audio frame will have no samples. If this frame goes through a signed linear translation path an interpolated frame can be created (if generic packet loss concealment is enabled) that has minimal data on it, including no src. If this frame is given to a translation path a crash will occur due to the lack of src. ASTERISK-28499 Change-Id: I024d10dd98207eb8a6b35b59880bcdf1090538f8	7 years ago
George Joseph	6407ccd2d9	dns_core: Create new API ast_dns_resolve_ipv6_and_ipv4 ... The new function takes in a pointer to an ast_sockaddr structure, a hostname and an optional port and then dispatches parallel "AAAA" and "A" record queries. If an "AAAA" record is returned, it's parsed into the ast_sockaddr structure along with the port if it was supplied. If no "AAAA" record was returned, the first "A" record returned (if any) is parsed instead. This is a synchronous call. If you need asynchronous lookups, use ast_dns_query_set_resolve_async and roll your own. Change-Id: I194b0b0e73da94b35cc35263a868ffac3a8d0a95	7 years ago
Sean Bright	51fd43206b	audiohook.c: Substitute silence for unavailable audio frames ... There are 4 scenarios to consider when capturing audio from a channel with an audiohook: 1. There is no rx and no tx audio, so return nothing. 2. There is rx but no tx audio, so return rx. 3. There is tx but no rx audio, so return tx. 4. There is rx and tx audio, so mix them and return. The file passed as the primary argument to MixMonitor will be written to in scenarios 2, 3, and 4. However, if you pass the r() and t() options to MixMonitor, a frame will only be written to the r() file if there was rx audio and a frame will only be written to the t() file if there was tx audio. If you subsequently take the r() and t() files and try to mix them, the sides of the conversation will 'drift' and be non-representative of the user experience. This patch adds a new 'S' option to MixMonitor that injects a frame of silence on either the r() side or the t() side of the channel so that when later mixed, there is no such drift. Change-Id: Ibf5ed73a811087727bd561a89a59f4447b4ee20e	7 years ago
George Joseph	c8c33c9a0b	Merge "srtp: Fix possible race condition, and add NULL checks" into 16	7 years ago
Kevin Harwell	d4766a82a2	srtp: Fix possible race condition, and add NULL checks ... Somehow it's possible for the srtp session object to be NULL even though the Asterisk srtp object itself is valid. When this happened it would cause a crash down in the srtp code when attempting to protect or unprotect data. After looking at the code there is at least one spot that makes this situation possible. If Asterisk fails to unprotect the data, and after several retries it still can't then the srtp->session gets freed, and set to NULL while still leaving the Asterisk srtp object around. However, according to the original issue reporter this does not appear to be their situation since they found no errors logged stating the above happened (which Asterisk does for that situation). An issue was found however, where a possible race condition could occur between the pjsip incoming negotiation, and the receiving of RTP packets. Both places could attempt to create/setup srtp for the same rtp instance at the same time. This potentially could be the cause of the problem as well. Given the above this patch adds locking around srtp setup for a given rtp, or rtcp instance. NULL checks for the session have also been added within the protect and unprotect functions as a precaution. These checks should at least stop Asterisk from crashing if it gets in this situation again. This patch also fixes one other issue noticed during investigation. When doing a replace the old object was freed before creating the replacement. If the new replacement object failed to create then the rtp/rtcp instance would now point to freed srtp data which could potentially cause a crash as well when the next attempt to reference it was made. This is now fixed so the old srtp object is kept upon replacement failure. Lastly, more logging has been added to help diagnose future issues. ASTERISK-28472 Change-Id: I240e11cbb1e9ea8083d59d50db069891228fe5cc	7 years ago
Joshua Colp	6350f4e278	cdr / cel: Use event time at event creation instead of processing. ... When updating times on CDR or CEL records using the time at which it is done can result in times being incorrect if the system is heavily loaded and stasis message processing is delayed. This change instead makes it so CDR and CEL use the time at which the stasis messages that drive the systems are created. This allows them to be backed up while still producing correct records. ASTERISK-28498 Change-Id: I6829227e67aefa318efe5e183a94d4a1b4e8500a	7 years ago
George Joseph	9feb13c5b1	Merge "various modules: json integer overflow" into 16	7 years ago
George Joseph	ef38087532	Merge "main/udptl.c: correctly handle udptl sequence wrap around" into 16	7 years ago
Kevin Harwell	6bb14150c4	various modules: json integer overflow ... There were still a few places in the code that could overflow when "packing" a json object with a value outside the base type integer's range. For instance: unsigned int value = INT_MAX + 1 ast_json_pack("{s: i}", value); would result in a negative number being "packed". In those situations this patch alters those values to a ast_json_int_t, which widens the value up to a long or long long. ASTERISK-28480 Change-Id: Ied530780d83e6f1772adba0e28d8938ef30c49a1	7 years ago
Friendly Automation	d3508f5b51	Merge "manager: Send fewer packets" into 16	7 years ago
Torrey Searle	83390327b2	main/udptl.c: correctly handle udptl sequence wrap around ... incorrect handling of UDPTL squence number wrap arounds causes loss of packets every time the wrap around occurs ASTERISK-28483 #close Change-Id: I33caeb2bf13c574a1ebb81714b58907091d64234	7 years ago
Sean Bright	0ebfc4a19d	manager: Send fewer packets ... The functions that build manager message headers do so in a way that results in a single messages being split across multiple packets. While this doesn't matter to the remote end, it makes network captures noisier and harder to follow, and also means additional system calls. With this patch, we build up more of the message content into the TLS buffer before flushing to the network. This change is completely internal to the manager code and does not affect any of the existing API's consumers. Change-Id: I50128b0769060ca5272dbbb5e60242d131eaddf9	7 years ago