6513 Commits (73641deedb38746b1f80a80f424ddca94d04b58f)

Author	SHA1	Message	Date
Kevin Harwell	c47dbf1fbe	message & stasis/messaging: make text message variables work in ARI ... When a text message was received any associated variable was not written to the ARI TextMessageReceived event. This occurred because Asterisk only wrote out "send" variables. However, even those "send" variables would fail ARI validation due to a TextMessageVariable formatting bug. Since it seems the TextMessageReceived event has never been able to include actual variables it was decided to remove the TextMessageVariable object type from ARI, and simply return a JSON object of key/value pairs for variables. This aligns more with how the ARI sendMessage handles variables, and other places in ARI. That being the case, and since this is technically an API breaking change (no one should really be affected since things never really worked) the ARI version was updated to reflect that. ASTERISK-28755 #close Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f	6 years ago
Kevin Harwell	d7bb1241ab	Merge "say: Remove unused "plural" option from main/say" into 17	6 years ago
Kevin Harwell	38e88f9469	Merge "format_cap: make function parameters 'const'" into 17	6 years ago
Kevin Harwell	31e9c110f5	Merge "pjsip: Update ACLs on named ACL changes." into 17	6 years ago
Walter Doekes	a3f5a80a59	say: Remove unused "plural" option from main/say ... There are exceptions for plural objects, but they are detected using the supplied NUMBER, not using an extra option. Change-Id: I95d1d1b2796b1aba92048a2dbae8a3856ed8a113	6 years ago
Kevin Harwell	d77a89cb9e	format_cap: make function parameters 'const' ... There were a couple places where the format cap function parameter was not 'const' when it should have been. This patch makes them 'const'. Change-Id: Ife753fb16a962d842a6b44f45363a61a66bfdb2e	6 years ago
George Joseph	fd31ef67b9	Merge "tcptls.c: Log more informative OpenSSL errors" into 17	6 years ago
George Joseph	e4f5142e23	Merge "bridging: Add better support for adding/removing streams." into 17	6 years ago
George Joseph	e300d57134	Merge "app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used" into 17	6 years ago
Joshua C. Colp	bc3c095e9f	pjsip: Update ACLs on named ACL changes. ... This change extends the Sorcery API to allow a wizard to be told to explicitly reload objects or a specific object type even if the wizard believes that nothing has changed. This has been leveraged by res_pjsip and res_pjsip_acl to reload endpoints and PJSIP ACLs when a named ACL changes. ASTERISK-28697 Change-Id: Ib8fee9bd9dd490db635132c479127a4114c1ca0b	6 years ago
Sean Bright	9ef514ae98	tcptls.c: Log more informative OpenSSL errors ... Dump OpenSSL's error stack to the error log when things fail. ASTERISK-28750 #close Reported by: Martin Zeh Change-Id: Ib63cd0df20275586e68ac4c2ddad222ed7bd9c0a	6 years ago
Joshua C. Colp	957681e08b	bridging: Add better support for adding/removing streams. ... This change adds support to bridge_softmix to allow the addition and removal of additional video source streams. When such a change occurs each participant is renegotiated as needed to reflect the update. If another video source is added then each participant gets another source. If a video source is removed then it is removed from each participant. This functionality allows you to have both your webcam and screenshare providing video if you desire, or even more streams. Mapping has been changed to use the topology index on the source channel as a unique identifier for outgoing participant streams, this will never change and provides an easy way to establish the mapping. The bridge_simple and bridge_native_rtp modules have also been updated to renegotiate when the stream topology of a party changes allowing the same behavior to occur as added to bridge_softmix. If a screen share is added then the opposite party is renegotiated. If that screen share is removed then the opposite party is renegotiated again. Some additional fixes are also included in here. Stream state is now conveyed in SDP so sendonly/recvonly/inactive streams can be requested. Removed streams now also remove previous state from themselves so consumers don't get confused. ASTERISK-28733 Change-Id: I93f41fb41b85646bef71408111c17ccea30cb0c5	6 years ago
Sean Bright	5f28ecc7af	app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used ... When opening a file for writing, Asterisk silently converts filenames ending with 'wav49' to 'WAV.' We aren't taking that in to account when setting the MIXMONITOR_FILENAME variable in MixMonitor. * If the user wants to write to a wav49 file, make sure that it is reflected properly in MIXMONITOR_FILENAME. * Add a note to the documentation describing this behavior. * Add a note in main/file.c indicating that app_mixmonitor needs to be changed if the logic in build_filename was changed. ASTERISK-24798 #close Reported by: xrobau Change-Id: I384691ce624eb55c80a125b9ca206d2d691c574c	6 years ago
Joshua C. Colp	5a0218a960	stasis: Use format specifier for size_t. ... Change-Id: Ic9b4afcc5398e7f46314419fc3c90433d818e35c	6 years ago
George Joseph	54cd865946	message.c: Add option to suppress the Message channel AMI and ARI events ... In order to reduce the amount of AMI and ARI events generated, the global "Message/ast_msg_queue" channel can be set to suppress it's normal channel housekeeping events such as "Newexten", "VarSet", etc. This can greatly reduce load on the manager and ARI applications when the Digium Phone Module for Asterisk is in use. To enable, set "hide_messaging_ami_events" in asterisk.conf to "yes" In Asterisk versions <18, the default is "no" preserving existing behavior. Beginning with Asterisk 18, the option will default to "yes". NOTE: This change does not affect UserEvents or the ARI TextMessageReceived events. * Added the "hide_messaging_ami_events" option to asterisk.conf. * Changed message.c to set the AST_CHAN_TP_INTERNAL property on the "Message/ast_msg_queue" channel if the option is set in asterisk.conf. This suppresses the reporting of the events. Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b	6 years ago
Friendly Automation	6747dd31b4	Merge "http: Add ability to disable /httpstatus URI" into 17	6 years ago
George Joseph	f34a4d5a77	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	6 years ago
Sean Bright	1e79bf44fc	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	6 years ago
Sean Bright	687890f734	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	6 years ago
Joshua Colp	7a9a4575bc	Merge "app_voicemail, say: Fix various leading whitespace problems" into 17	6 years ago
Sean Bright	cb6f106d76	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	6 years ago
Sean Bright	9ff062f994	pbx.c: Include filesystem cache in free memory calculation ... ASTERISK-28695 #close Reported by: Kevin Flyn Change-Id: Ief098bb6eb77378daeace8f97ba30701c8de55b8	6 years ago
Friendly Automation	e98856b5c7	Merge "res_pjsip_endpoint_identifier_ip.c: Add port matching support" into 17	6 years ago
Friendly Automation	21334e00c5	Merge "stasis.c: Use correct topic name in stasis_topic_pool_delete_topic" into 17	6 years ago
Sean Bright	eac6eb663d	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	6 years ago
Friendly Automation	b2a7d6ccfb	Merge "features.c: Make Bridge application tolerate unspecified channel." into 17	6 years ago
George Joseph	e0449939ec	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	6 years ago
Richard Mudgett	919da21690	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	6 years ago
Sean Bright	f344356290	websocket: Consider pending SSL data when waiting for socket input ... When TLS is in use, checking the readiness of the underlying FD is insufficient for determining if there is data available to be read. So before polling the FD, check if there is any buffered data in the TLS layer and use that first. ASTERISK-28562 #close Reported by: Robert Sutton Change-Id: I95fcb3e2004700d5cf8e5ee04943f0115b15e10d	6 years ago
Sean Bright	843d601e2c	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
George Joseph	86550a59f3	Merge "config.c: Skip UTF-8 BOMs if present when reading config files" into 17	6 years ago
Joshua C. Colp	7aa2445264	Merge "confbridge: Add support for specifying maximum sample rate." into 17	6 years ago
Sean Bright	8c8989f205	config.c: Skip UTF-8 BOMs if present when reading config files ... ASTERISK-28667 #close Change-Id: I4767ed365c98f3e1587b7653321048a31d8a53b2	6 years ago
Kevin Reeves	45f7f9bb85	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	6 years ago
Joshua C. Colp	41fdcf7a65	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
Jaco Kroon	6041fc8b9c	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	2526c8e562	Merge "channel.c: Resolve issue with receiving SIP INFO packets for DTMF" into 17	6 years ago
George Joseph	88150323a2	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
Sean Bright	cc59e21409	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	6 years ago
George Joseph	6b1ba58967	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
Friendly Automation	43a7e44fbe	Merge "serializer: set high/low alert levels on whole pool" into 17	6 years ago
Alexei Gradinari	649733612d	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	8bc6fa0fbd	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	bbf995320e	Merge "pbx: deadlock when outgoing dialed channel hangs up too quickly" into 17	6 years ago
Kevin Harwell	ae761c7473	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	37ec88c4c8	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
Corey Farrell	25fbe79793	stasis_state: Create internal stasis_state_proxy object. ... This improves the way which stasis_state reference counting works. Since manager->states holds onto the proxy object instead of the real object this allows stasis_state objects to be freed when appropriate without use of a special state_remove function. Additionally each distinct eid associated with the state holds a reference to the state to prevent early release and potentially allow easier debug of leaks. Change-Id: I400e0db4b9afa3d5cb4ac7dad60907897e73f9a9	6 years ago
Joshua Colp	57fa604571	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	6 years ago
George Joseph	23858b7ef4	Merge "pbx: Prevent Realtime switch crash on invalid priority" into 17	6 years ago
Sean Bright	8d0edf2b37	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