6030 Commits (dafdcc623cd46dba3663a3b5bb2663f8630a3a00)

Author	SHA1	Message	Date
Friendly Automation	b0b229f63c	Merge "http: Add ability to disable /httpstatus URI" into 13	5 years ago
George Joseph	4eb2ad5e17	cdr.c: Set event time on party b when leaving a parking bridge ... When Alice calls Bob and Bob does a blind transfer to Charlie, Bob's bridge leave event generates a finalize on both the party_a and party_b CDRs but while the party_a CDR has the correct end time set from the event time, party_b's leg did not. This caused that CDR's end time to be equal to the answered time and resulted in a billsec of 0. * We now pass the bridge leave message event time to cdr_object_party_b_left_bridge_cb() and set it on that CDR before calling cdr_object_finalize() on it. NOTE: This issue affected transfers using chan_sip most of the time but also occasionally affected chan_pjsip probably due to message timing. ASTERISK-28677 Reported by: Maciej Michno Change-Id: I790720f1e7326f9b8ce8293028743b0ef0fb2cca	5 years ago
Sean Bright	b6ebdabd9c	http: Add ability to disable /httpstatus URI ... Add a new configuration option 'enable_status' which allows the /httpstatus URI handler to be administratively disabled. We also no longer unconditionally register the /static and /httpstatus URI handlers, but instead do it based upon configuration. Behavior change: If enable_static was turned off, the URI handler was still installed but returned a 403 when it was accessed. Because we now register/unregister the URI handlers as appropriate, if the /static URI is disabled we will return a 404 instead. Additionally: * Change 'enablestatic' to 'enable_static' but keep the former for backwards compatibility. * Improve some internal variable names ASTERISK-28710 #close Change-Id: I647510f796473793b1d3ce1beb32659813be69e1	5 years ago
Sean Bright	ca7309fd29	translate.c: Fix silk 24kHz truncation in 'core show translation' ... SILK @ 24kHz is not shown in the 'core show translation' output because of an off-by-one-error. Discovered while looking into ASTERISK~19871. ASTERISK-28706 Reported by: Sean Bright Change-Id: Ie1a551a8a484e07b45c8699cc0c90f1061029510	5 years ago
Joshua Colp	073d67ece0	Merge "app_voicemail, say: Fix various leading whitespace problems" into 13	5 years ago
Sean Bright	ee05a43951	app_voicemail, say: Fix various leading whitespace problems ... In af90afd90c, Japanese language support was added to app_voicemail and main/say.c, but the leading whitespace is not consistent with Asterisk coding guidelines. This patch fixes that. Whitespace only, no functional change. ASTERISK~23324 Reported by: Kevin McCoy Change-Id: I72c725f5930084673749bd7c9cc426a987f08e87	5 years ago
Sean Bright	884b86e7c0	pbx.c: Include filesystem cache in free memory calculation ... ASTERISK-28695 #close Reported by: Kevin Flyn Change-Id: Ief098bb6eb77378daeace8f97ba30701c8de55b8	5 years ago
Joshua Colp	4cefb2f3cb	Merge "res_pjsip_endpoint_identifier_ip.c: Add port matching support" into 13	5 years ago
Joshua Colp	379050504d	Merge "stasis.c: Use correct topic name in stasis_topic_pool_delete_topic" into 13	5 years ago
Sean Bright	49bb7d85cd	res_pjsip_endpoint_identifier_ip.c: Add port matching support ... Adds source port matching support when IP matching is used: [example] type = identify match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444 If the IP matches but the source port does not, we reject and search for alternatives. SRV lookups are still performed if enabled (srv_lookups = yes), unless the configured FQDN includes a port number in which case just a host lookup is performed. ASTERISK-28639 #close Reported by: Mitch Claborn Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92	5 years ago
George Joseph	e4e5e07e3d	stasis.c: Use correct topic name in stasis_topic_pool_delete_topic ... When a topic is created for an object, its name is only <object>:<uniqueid> For example: bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 When a topic is added to a pool, its name has the pool's topic name prepended. For example: bridge:all/bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 The topic_pool_entry's name however, is only what was passed in to stasis_topic_pool_get_topic which is bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 That's actually correct because the entry is qualified by the pool that's in. When you're ready to delete the entry from the pool, you retrieve the tropic name from the object but since it now has the pool's topic name prepended, it won't be found in the pool container. Fix: * Modified stasis_topic_pool_delete_topic() to skip past the pool topic's name, if it was prepended to the topic name, before searching the container for a pool entry. ASTERISK-28633 Reported by: Joeran Vinzens Change-Id: I4396aa69dd83e4ab84c5b91b39293cfdbcf483e6	5 years ago
Richard Mudgett	9997710f8f	features.c: Make Bridge application tolerate unspecified channel. ... The Bridge application was inconsistent if the channel to bridge with is not specified. If no parameters are given then a warning is issued and the current channel is hung up. If options are given but no channel is specified then a warning is issued and the current channel is not hung up. * Made the Bridge application give a verbose message instead of a warning if the channel to bridge with is not specified and made not hang up the current channel. As a result dialplan no longer needs to check if a channel name is passed before calling Bridge and simply needs to check the BRIDGERESULT channel variable instead. This is something you likely want your dialplan to do anyway. * Fixed up L() option warning message. It is up to the caller to determine if the channel is hung up because of the warning. Dial() hangs up the current channel while Bridge() does not. Change-Id: I44349a8dc3912397f28852777de04f19e7bb9c73	5 years ago
Sean Bright	401056fd27	db: Initialize condition primitive before use ... The db_init() function ultimately calls db_sync() which signals the condition before it is initialized. Change-Id: Id4a4e025b637bc4ac7d90557fcb71d56598892ab	6 years ago
Friendly Automation	1a96383e29	Merge "config.c: Skip UTF-8 BOMs if present when reading config files" into 13	6 years ago
Friendly Automation	66970af0d3	Merge "confbridge: Add support for specifying maximum sample rate." into 13	6 years ago
Sean Bright	6ceb4c25fb	config.c: Skip UTF-8 BOMs if present when reading config files ... ASTERISK-28667 #close Change-Id: I4767ed365c98f3e1587b7653321048a31d8a53b2	6 years ago
Friendly Automation	f7fcb601ab	Merge "json: Support older jansson versions." into 13	6 years ago
Joshua C. Colp	c1faa9ecda	json: Support older jansson versions. ... The use of '?' is a fairly new addition to jansson and is not supported in the version of jansson that can be used by 13. This change returns to previous supported behavior and removes usage of '?'. ASTERISK-28663 Change-Id: I6d596007ae85e8724d928865d99968f679be1142	6 years ago
Joshua C. Colp	ca35de4282	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	6 years ago
Joshua Colp	089b7529df	Merge "ACL: ast_apply_acl_nolog - identical to ast_apply_acl but without logging." into 13	6 years ago
Jaco Kroon	6e79028020	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>	6 years ago
George Joseph	5aee4f7655	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	6 years ago
George Joseph	1b9281a5de	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	6 years ago
Alexei Gradinari	26fc3a3127	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	6 years ago
Kevin Harwell	6ca76798e1	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	6 years ago
Friendly Automation	59487acfb8	Merge "pbx: deadlock when outgoing dialed channel hangs up too quickly" into 13	6 years ago
Kevin Harwell	a2ddaaac32	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	6 years ago
Kevin Harwell	0b7efc1687	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	6 years ago
George Joseph	c6f81a7ba4	Merge "taskprocessor.c: Added "like" support to 'core show taskprocessors'" into 13	6 years ago
Ben Ford	67bd02d08d	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	6 years ago
Sean Bright	7280e2ec51	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	6 years ago
Friendly Automation	fde6ed7d91	Merge "core: Fix ABI mismatch of ao2_global_obj." into 13	6 years ago
George Joseph	c5a9c039ba	Merge "taskprocessor.c: Add CLI commands to reset taskprocessor stats." into 13	6 years ago
Corey Farrell	c881d36319	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	6 years ago
Ben Ford	3069153bdf	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	6 years ago
Friendly Automation	d6c52479a4	Merge "astmm.c: Display backtrace with memory show allocations" into 13	6 years ago
George Joseph	afd6b98757	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	6 years ago
Corey Farrell	48fd773743	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	6 years ago
Sean Bright	622e487cd4	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	6 years ago
Joshua Colp	c711546559	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	6 years ago
Sean Bright	ddc64ca059	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	6 years ago
George Joseph	7251b76c62	Merge "srtp: Fix possible race condition, and add NULL checks" into 13	6 years ago
Friendly Automation	5598dc045b	Merge "cdr / cel: Use event time at event creation instead of processing." into 13	6 years ago
Kevin Harwell	feaadbd250	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	6 years ago
Joshua Colp	1496aef51b	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	6 years ago
George Joseph	a57b59c959	Merge "various modules: json integer overflow" into 13	6 years ago
Torrey Searle	6f77e12f21	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	6 years ago
Kevin Harwell	f145b58542	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	6 years ago
Sean Bright	5096a5ddf7	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	6 years ago
George Joseph	8d1acc27d5	Merge "sched: Don't allow ast_sched_del to deadlock ast_sched_runq from same thread" into 13	6 years ago