6854 Commits (22.0.0-pre1)

Author	SHA1	Message	Date
George Joseph	2328a0ab3a	manager.c: Fix FRACK when doing CoreShowChannelMap in DEVMODE ... If you run an AMI CoreShowChannelMap on a channel that isn't in a bridge and you're in DEVMODE, you can get a FRACK because the bridge id is empty. We now simply return an empty list for that request.	1 year ago
Ben Ford	fb0e46e464	channel: Add multi-tenant identifier. ... This patch introduces a new identifier for channels: tenantid. It's a stringfield on the channel that can be used for general purposes. It will be inherited by other channels the same way that linkedid is. You can set tenantid in a few ways. The first is to set it in the dialplan with the Set and CHANNEL functions: exten => example,1,Set(CHANNEL(tenantid)=My tenant ID) It can also be accessed via CHANNEL: exten => example,2,NoOp(CHANNEL(tenantid)) Another method is to use the new tenantid option for pjsip endpoints in pjsip.conf: [my_endpoint] type=endpoint tenantid=My tenant ID This is considered the best approach since you will be able to see the tenant ID as early as the Newchannel event. It can also be set using set_var in pjsip.conf on the endpoint like setting other channel variable: set_var=CHANNEL(tenantid)=My tenant ID Note that set_var will not show tenant ID on the Newchannel event, however. Tenant ID has also been added to CDR. It's read-only and can be accessed via CDR(tenantid). You can also get the tenant ID of the last channel communicated with via CDR(peertenantid). Tenant ID will also show up in CEL records if it has been set, and the version number has been bumped accordingly. Fixes: #740 UserNote: tenantid has been added to channels. It can be read in dialplan via CHANNEL(tenantid), and it can be set using Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to use the new tenantid option for pjsip endpoints (e.g., tenantid=My tenant ID) so that it will show up in Newchannel events. You can set it like any other channel variable using set_var in pjsip.conf as well, but note that this will NOT show up in Newchannel events. Tenant ID is also available in CDR and can be accessed with CDR(tenantid). The peer tenant ID can also be accessed with CDR(peertenantid). CEL includes tenant ID as well if it has been set. UpgradeNote: A new versioned struct (ast_channel_initializers) has been added that gets passed to __ast_channel_alloc_ap. The new function ast_channel_alloc_with_initializers should be used when creating channels that require the use of this struct. Currently the only value in the struct is for tenantid, but now more fields can be added to the struct as necessary rather than the __ast_channel_alloc_ap function. A new option (tenantid) has been added to endpoints in pjsip.conf as well. CEL has had its version bumped to include tenant ID.	1 year ago
gibbz00	99a5064a07	feat: ARI "ChannelToneDetected" event ... A stasis event is now produced when using the TONE_DETECT dialplan function. This event is published over ARI using the ChannelToneDetected event. This change does not make it available over AMI. Fixes: #811 UserNote: Setting the TONE_DETECT dialplan function on a channel in ARI will now cause a ChannelToneDetected ARI event to be raised when the specified tone is detected.	1 year ago
George Joseph	0f67c2696f	manager.c: Add entries to Originate blacklist ... Added Reload and DBdeltree to the list of dialplan application that can't be executed via the Originate manager action without also having write SYSTEM permissions. Added CURL, DB*, FILE, ODBC and REALTIME* to the list of dialplan functions that can't be executed via the Originate manager action without also having write SYSTEM permissions. If the Queue application is attempted to be run by the Originate manager action and an AGI parameter is specified in the app data, it'll be rejected unless the manager user has either the AGI or SYSTEM permissions. Resolves: #GHSA-c4cg-9275-6w44	1 year ago
George Joseph	cee50bab7a	rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset() ... There can be empty slots in payload_mapping_tx corresponding to dynamic payload types that haven't been seen before so we now check for NULL before attempting to use 'type' in the call to ast_format_cmp. Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset() Resolves: #822	1 year ago
George Joseph	b8c440a52f	stasis_channels: Use uniqueid and name to delete old snapshots ... Whenver a new channel snapshot is created or when a channel is destroyed, we need to delete any existing channel snapshot from the snapshot cache. Historically, we used the channel->snapshot pointer to delete any existing snapshots but this has two issues. First, if something (possibly ast_channel_internal_swap_snapshots) sets channel->snapshot to NULL while there's still a snapshot in the cache, we wouldn't be able to delete it and it would be orphaned when the channel is destroyed. Since we use the cache to list channels from the CLI, AMI and ARI, it would appear as though the channel was still there when it wasn't. Second, since there are actually two caches, one indexed by the channel's uniqueid, and another indexed by the channel's name, deleting from the caches by pointer requires a sequential search of all of the hash table buckets in BOTH caches to find the matching snapshots. Not very efficient. So, we now delete from the caches using the channel's uniqueid and name. This solves both issues. This doesn't address how channel->snapshot might have been set to NULL in the first place because although we have concrete evidence that it's happening, we haven't been able to reproduce it. Resolves: #783	1 year ago
George Joseph	6e73af341a	app_voicemail_odbc: Allow audio to be kept on disk ... This commit adds a new voicemail.conf option 'odbc_audio_on_disk' which when set causes the ODBC variant of app_voicemail to leave the message and greeting audio files on disk and only store the message metadata in the database. This option came from a concern that the database could grow to large and cause remote access and/or replication to become slow. In a clustering situation with this option, all asterisk instances would share the same database for the metadata and either use a shared filesystem or other filesystem replication service much more suitable for synchronizing files. The changes to app_voicemail to implement this feature were actually quite small but due to the complexity of the module, the actual source code changes were greater. They fall into the following categories: * Tracing. The module is so complex that it was impossible to figure out the path taken for various scenarios without the addition of many SCOPE_ENTER, SCOPE_EXIT and ast_trace statements, even in code that's not related to the functional change. Making this worse was the fact that many "if" statements in this module didn't use braces. Since the tracing macros add multiple statements, many "if" statements had to be converted to use braces. * Excessive use of PATH_MAX. Previous maintainers of this module used PATH_MAX to allocate character arrays for filesystem paths and SQL statements as though they cost nothing. In fact, PATH_MAX is defined as 4096 bytes! Some functions had (and still have) multiples of these. One function has 7. Given that the vast majority of installations use the default spool directory path `/var/spool/asterisk/voicemail`, the actual path length is usually less than 80 bytes. That's over 4000 bytes wasted. It was the same for SQL statement buffers. A 4K buffer for statement that only needed 60 bytes. All of these PATH_MAX allocations in the ODBC related code were changed to dynamically allocated buffers. The rest will have to be addressed separately. * Bug fixes. During the development of this feature, several pre-existing ODBC related bugs were discovered and fixed. They had to do with leaving orphaned files on disk, not preserving original message ids when moving messages between folders, not honoring the "formats" config parameter in certain circumstances, etc. UserNote: This commit adds a new voicemail.conf option 'odbc_audio_on_disk' which when set causes the ODBC variant of app_voicemail_odbc to leave the message and greeting audio files on disk and only store the message metadata in the database. Much more information can be found in the voicemail.conf.sample file.	1 year ago
Tinet-mucw	7d28165cb1	bridge_basic.c: Make sure that ast_bridge_channel is not destroyed while iterating over bridge->channels. ... From the gdb information, we can see that while iterating over bridge->channels, the ast_bridge_channel reference count is 0, indicating it has already been destroyed.Additionally, when ast_bridge_channel is removed from bridge->channels, the bridge is first locked. Therefore, locking the bridge before iterating over bridge->channels can resolve the race condition. Resolves: https://github.com/asterisk/asterisk/issues/768	1 year ago
Alexei Gradinari	4280d7cd77	pbx.c: expand fields width of "core show hints" ... The current width for "extension" is 20 and "device state id" is 20, which is too small. The "extension" field contains "ext"@"context", so 20 characters is not enough. The "device state id" field, for example for Queue pause state contains Queue:"queue_name"_pause_PSJIP/"endpoint", so the 20 characters is not enough. Increase the width of "extension" field to 30 characters and the width of the "device state id" field to 60 characters. Resolves: #770 UserNote: The fields width of "core show hints" were increased. The width of "extension" field to 30 characters and the width of the "device state id" field to 60 characters.	1 year ago
Sean Bright	26eabed246	manager.c: Properly terminate `CoreShowChannelMap` event. ... Fixes #761	1 year ago
Bastian Triller	843cc8f2cd	cli: Show configured cache dir ... Since Asterisk 19 it is possible to cache recorded files into another directory [1] [2]. Show configured location of cache dir in CLI's core show settings. [1] ASTERISK-29143 [2] b08427134f	1 year ago
Sean Bright	f3e88d366c	xml.c: Update deprecated libxml2 API usage. ... Two functions are deprecated as of libxml2 2.12: * xmlSubstituteEntitiesDefault * xmlParseMemory So we update those with supported API. Additionally, `res_calendar_caldav` has been updated to use libxml2's xmlreader API instead of the SAX2 API which has always felt a little hacky (see deleted comment block in `res_calendar_caldav.c`). The xmlreader API has been around since libxml2 2.5.0 which was released in 2003. Fixes #725	1 year ago
Sean Bright	e1db3027af	asterisk.c: Don't log an error if .asterisk_history does not exist. ... Fixes #751	1 year ago
Sean Bright	f3449f3370	file.h: Rename function argument to avoid C++ keyword clash. ... Fixes #744	1 year ago
Mike Bradeen	3b4d2a6506	rtp_engine: add support for multirate RFC2833 digits ... Add RFC2833 DTMF support for 16K, 24K, and 32K bitrate codecs. Asterisk currently treats RFC2833 Digits as a single rtp payload type with a fixed bitrate of 8K. This change would expand that to 8, 16, 24 and 32K. This requires checking the offered rtp types for any of these bitrates and then adding an offer for each (if configured for RFC2833.) DTMF generation must also be changed in order to look at the current outbound codec in order to generate appropriately timed rtp. For cases where no outgoing audio has yet been sent prior to digit generation, Asterisk now has a concept of a 'preferred' codec based on offer order. On inbound calls Asterisk will mimic the payload types of the RFC2833 digits. On outbound calls Asterisk will choose the next free payload types starting with 101. UserNote: No change in configuration is required in order to enable this feature. Endpoints configured to use RFC2833 will automatically have this enabled. If the endpoint does not support this, it should not include it in the SDP offer/response. Resolves: #699	1 year ago
Ivan Poddubny	29675e3aab	asterisk.c: Fix sending incorrect messages to systemd notify ... Send "RELOADING=1" instead of "RELOAD=1" to follow the format expected by systemd (see sd_notify(3) man page). Do not send STOPPING=1 in remote console mode: attempting to execute "asterisk -rx" by the main process leads to a warning if NotifyAccess=main (the default) or to a forced termination if NotifyAccess=all.	1 year ago
Naveen Albert	1f1c5bb902	logger: Add unique verbose prefixes for levels 5-10. ... Add unique verbose prefixes for levels higher than 4, so that these can be visually differentiated from each other. Resolves: #721	2 years ago
Naveen Albert	7cffbabe5c	say.c: Fix cents off-by-one due to floating point rounding. ... Some of the money announcements can be off by one cent, due to the use of floating point in the money calculations, which is bad for obvious reasons. This replaces floating point with simple string parsing to ensure the cents value is converted accurately. Resolves: #525	2 years ago
Naveen Albert	01ff5c8140	loader.c: Allow dependent modules to be unloaded recursively. ... Because of the (often recursive) nature of module dependencies in Asterisk, hot swapping a module on the fly is cumbersome if a module is depended on by other modules. Currently, dependencies must be popped manually by unloading dependents, unloading the module of interest, and then loading modules again in reverse order. To make this easier, the ability to do this recursively in certain circumstances has been added, as an optional extension to the "module refresh" command. If requested, Asterisk will check if a module that has a positive usecount could be unloaded safely if anything recursively dependent on it were unloaded. If so, it will go ahead and unload all these modules and load them back again. This makes hot swapping modules that provide dependencies much easier. Resolves: #474 UserNote: In certain circumstances, modules with dependency relations can have their dependents automatically recursively unloaded and loaded again using the "module refresh" CLI command or the ModuleLoad AMI command.	2 years ago
George Joseph	8e119a72f0	tcptls/iostream: Add support for setting SNI on client TLS connections ... If the hostname field of the ast_tcptls_session_args structure is set (which it is for websocket client connections), that hostname will now automatically be used in an SNI TLS extension in the client hello. Resolves: #713 UserNote: Secure websocket client connections now send SNI in the TLS client hello.	2 years ago
Spiridonov Dmitry	7e7a603360	sorcery.c: Fixed crash error when executing "module reload" ... Fixed crash error when cli "module reload". The error appears when compiling with res_prometheus and using the sorcery memory cache for registrations	2 years ago
Naveen Albert	c1b69d460d	callerid.c: Parse previously ignored Caller ID parameters. ... Commit f2f397c1a8 previously made it possible to send Caller ID parameters to FXS stations which, prior to that, could not be sent. This change is complementary in that we now handle receiving all these parameters on FXO lines and provide these up to the dialplan, via chan_dahdi. In particular: * If a redirecting reason is provided, the channel's redirecting reason is set. No redirecting number is set, since there is no parameter for this in the Caller ID protocol, but the reason can be checked to determine if and why a call was forwarded. * If the Call Qualifier parameter is received, the Call Qualifier variable is set. * Some comments have been added to explain why some of the code is the way it is, to assist other people looking at it. With this change, Asterisk's Caller ID implementation is now reasonably complete for both FXS and FXO operation. Resolves: #681	2 years ago
Naveen Albert	a0b579807c	file.c, channel.c: Don't emit warnings if progress received. ... Silently ignore AST_CONTROL_PROGRESS where appropriate, as most control frames already are. Resolves: #696	2 years ago
George Joseph	70fd8f1c93	rtp_engine and stun: call ast_register_atexit instead of ast_register_cleanup ... rtp_engine.c and stun.c were calling ast_register_cleanup which is skipped if any loadable module can't be cleanly unloaded when asterisk shuts down. Since this will always be the case, their cleanup functions never get run. In a practical sense this makes no difference since asterisk is shutting down but if you're in development mode and trying to use the leak sanitizer, the leaks from both of those modules clutter up the output.	2 years ago
George Joseph	52881ef6e5	manager.c: Add missing parameters to Login documentation ... * Added the AuthType and Key parameters for MD5 authentication. * Added the Events parameter. Resolves: #689	2 years ago
George Joseph	f6b9d9e7d7	Fix incorrect application and function documentation references ... There were a few references in the embedded documentation XML where the case didn't match or where the referenced app or function simply didn't exist any more. These were causing 404 responses in docs.asterisk.org.	2 years ago
Sean Bright	2b5673b68d	cli.c: `core show channels concise` is not really deprecated. ... Fixes #675	2 years ago
jonatascalebe	988986bdc9	manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action ... manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action The action originate does not has the ability to run an subroutine at initial channel, like the Aplication Originate. This update give this ability for de action originate too. For example, we can run a routine via Gosub on the channel to request an automatic answer, so the caller does not need to accept the call when using the originate command via manager, making the operation more efficient. UserNote: When using the Originate AMI Action, we now can pass the PreDialGoSub parameter, instructing the asterisk to perform an subrouting at channel before call start. With this parameter an call initiated by AMI can request the channel to start the call automaticaly, adding a SIP header to using GoSUB, instructing to autoanswer the channel, and proceeding the outbuound extension executing. Exemple of an context to perform the previus indication: [addautoanswer] exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0) exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0) exten => _s,n,Return()	2 years ago
Naveen Albert	fc80bed5a7	pbx_variables.c: Prevent SEGV due to stack overflow. ... It is possible for dialplan to result in an infinite recursion of variable substitution, which eventually leads to stack overflow. If we detect this, abort substitution and log an error for the user to fix the broken dialplan. Resolves: #480 UpgradeNote: The maximum amount of dialplan recursion using variable substitution (such as by using EVAL_EXTEN) is capped at 15.	2 years ago
Naveen Albert	ef7788e0e4	manager.c: Add CLI command to kick AMI sessions. ... This adds a CLI command that can be used to manually kick specific AMI sessions. Resolves: #485 UserNote: The "manager kick session" CLI command now allows kicking a specified AMI session.	2 years ago
George Joseph	628f8d7a43	Stir/Shaken Refactor ... Why do we need a refactor? The original stir/shaken implementation was started over 3 years ago when little was understood about practical implementation. The result was an implementation that wouldn't actually interoperate with any other stir-shaken implementations. There were also a number of stir-shaken features and RFC requirements that were never implemented such as TNAuthList certificate validation, sending Reason headers in SIP responses when verification failed but we wished to continue the call, and the ability to send Media Key(mky) grants in the Identity header when the call involved DTLS. Finally, there were some performance concerns around outgoing calls and selection of the correct certificate and private key. The configuration was keyed by an arbitrary name which meant that for every outgoing call, we had to scan the entire list of configured TNs to find the correct cert to use. With only a few TNs configured, this wasn't an issue but if you have a thousand, it could be. What's changed? * Configuration objects have been refactored to be clearer about their uses and to fix issues. * The "general" object was renamed to "verification" since it contains parameters specific to the incoming verification process. It also never handled ca_path and crl_path correctly. * A new "attestation" object was added that controls the outgoing attestation process. It sets default certificates, keys, etc. * The "certificate" object was renamed to "tn" and had it's key change to telephone number since outgoing call attestation needs to look up certificates by telephone number. * The "profile" object had more parameters added to it that can override default parameters specified in the "attestation" and "verification" objects. * The "store" object was removed altogther as it was never implemented. * We now use libjwt to create outgoing Identity headers and to parse and validate signatures on incoming Identiy headers. Our previous custom implementation was much of the source of the interoperability issues. * General code cleanup and refactor. * Moved things to better places. * Separated some of the complex functions to smaller ones. * Using context objects rather than passing tons of parameters in function calls. * Removed some complexity and unneeded encapsuation from the config objects. Resolves: #351 Resolves: #46 UserNote: Asterisk's stir-shaken feature has been refactored to correct interoperability, RFC compliance, and performance issues. See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information. UpgradeNote: The stir-shaken refactor is a breaking change but since it's not working now we don't think it matters. The stir_shaken.conf file has changed significantly which means that existing ones WILL need to be changed. The stir_shaken.conf.sample file in configs/samples/ has quite a bit more information. This is also an ABI breaking change since some of the existing objects needed to be changed or removed, and new ones added. Additionally, if res_stir_shaken is enabled in menuselect, you'll need to either have the development package for libjwt v1.15.3 installed or use the --with-libjwt-bundled option with ./configure.	2 years ago
Sebastian Jennen	ea8ead4e13	translate.c: implement new direct comp table mode ... The new mode lists for each codec translation the actual real cost in cpu microseconds per second translated audio. This allows to compare the real cpu usage of translations and helps in evaluation of codec implementation changes regarding performance (regression testing). - add new table mode - hide the 999999 comp values, as these only indicate an issue with transcoding - hide the 0 values, as these also do not contain any information (only indicate a multistep transcoding) Resolves: #601	2 years ago
Naveen Albert	526a6e0ce4	dsp.c: Fix and improve potentially inaccurate log message. ... If ast_dsp_process is called with a codec besides slin, ulaw, or alaw, a warning is logged that in-band DTMF is not supported, but this message is not always appropriate or correct, because ast_dsp_process is much more generic than just DTMF detection. This logs a more generic message in those cases, and also improves codec-mismatch logging throughout dsp.c by ensuring incompatible codecs are printed out. Resolves: #595	2 years ago
George Joseph	29a273618d	pjsip show channelstats: Prevent possible segfault when faxing ... Under rare circumstances, it's possible for the original audio session in the active_media_state default_session to be corrupted instead of removed when switching to the t38/image media session during fax negotiation. This can cause a segfault when a "pjsip show channelstats" attempts to print that audio media session's rtp statistics. In these cases, the active_media_state topology is correctly showing only a single t38/image stream so we now check that there's an audio stream in the topology before attempting to use the audio media session to get the rtp statistics. Resolves: #592	2 years ago
George Joseph	6871d1cdfc	Reduce startup/shutdown verbose logging ... When started with a verbose level of 3, asterisk can emit over 1500 verbose message that serve no real purpose other than to fill up logs. When asterisk shuts down, it emits another 1100 that are of even less use. Since the testsuite runs asterisk with a verbose level of 3, and asterisk starts and stops for every one of the 700+ tests, the number of log messages is staggering. Besides taking up resources, it also makes it hard to debug failing tests. This commit changes the log level for those verbose messages to 5 instead of 3 which reduces the number of log messages to only a handful. Of course, NOTICE, WARNING and ERROR message are unaffected. There's also one other minor change... ast_context_remove_extension_callerid2() logs a DEBUG message instead of an ERROR if the extension you're deleting doesn't exist. The pjsip_config_wizard calls that function to clean up the config and has been triggering that annoying error message for years. Resolves: #582	2 years ago
Joshua C. Colp	edf54951be	utils: Make behavior of ast_strsep* match strsep. ... Given the scenario of passing an empty string to the ast_strsep functions the functions would return NULL instead of an empty string. This is counter to how strsep itself works. This change alters the behavior of the functions to match that of strsep. Fixes: #565	2 years ago
Brad Smith	b0992fb771	main/utils: Simplify the FreeBSD ast_get_tid() handling ... FreeBSD has had kernel threads for 20+ years.	2 years ago
Sean Bright	f2ac526172	rtp_engine.c: Correct sample rate typo for L16/44100. ... Fixes #555	2 years ago
Naveen Albert	f1a9ec4703	manager.c: Fix erroneous reloads in UpdateConfig. ... Currently, a reload will always occur if the Reload header is provided for the UpdateConfig action. However, we should not be doing a reload if the header value has a falsy value, per the documentation, so this makes the reload behavior consistent with the existing documentation. Resolves: #551	2 years ago
Naveen Albert	ac71e40042	logger: Fix linking regression. ... Commit 008731b0a4 caused a regression by resulting in logger.xml being compiled and linked into the asterisk binary in lieu of logger.c on certain platforms if Asterisk was compiled in dev mode. To fix this, we ensure the file has a unique name without the extension. Most existing .xml files have been named differently from any .c files in the same directory or did not pose this issue. channels/pjsip/dialplan_functions.xml does not pose this issue but is also being renamed to adhere to this policy. Resolves: #539	2 years ago
Naveen Albert	5cf75f9b2c	func_frame_trace: Add CLI command to dump frame queue. ... This adds a simple CLI command that can be used for analyzing all frames currently queued to a channel. A couple log messages are also adjusted to be more useful in tracing bridging problems. Resolves: #533	2 years ago
George Joseph	09052bfa51	Revert "core & res_pjsip: Improve topology change handling." ... This reverts commit 315eb551db. Over the past year, we've had several reports of "topology storms" occurring where 2 external facing channels connected by one or more local channels and bridges will get themselves in a state where they continually send each other topology change requests. This usually manifests itself in no-audio calls and a flood of "Exceptionally long queue length" messages. It appears that this commit is the cause so we're reverting it for now until we can determine a more appropriate solution. Resolves: #530	2 years ago
Naveen Albert	fa3922a4d2	manager.c: Fix regression due to using wrong free function. ... Commit 424be34563 introduced a regression by calling ast_free on memory allocated by realpath. This causes Asterisk to abort when executing this function. Since the memory is allocated by glibc, it should be freed using ast_std_free. Resolves: #513	2 years ago
Ben Ford	424be34563	manager.c: Prevent path traversal with GetConfig. ... When using AMI GetConfig, it was possible to access files outside of the Asterisk configuration directory by using filenames with ".." and "./" even while live_dangerously was not enabled. This change resolves the full path and ensures we are still in the configuration directory before attempting to access the file.	2 years ago
Naveen Albert	183954bed3	config_options.c: Fix truncation of option descriptions. ... This increases the format width of option descriptions to avoid needless truncation for longer descriptions. Resolves: #428	2 years ago
Naveen Albert	ce1f4b3018	manager.c: Improve clarity of "manager show connected". ... Improve the "manager show connected" CLI command to clarify that the last two columns are permissions related, not counts, and use sufficient widths to consistently display these values. ASTERISK-30143 #close Resolves: #482	2 years ago
Naveen Albert	d1fb397cfc	general: Fix broken links. ... This fixes a number of broken links throughout the tree, mostly caused by wiki.asterisk.org being replaced with docs.asterisk.org, which should eliminate the need for sporadic fixes as in f28047db36. Resolves: #430	2 years ago
Sean Bright	008731b0a4	logger.c: Move LOG_GROUP documentation to dedicated XML file. ... The `get_documentation` awk script will only extract the first DOCUMENTATION block that it finds in a given file. This is by design (9bc2127) to prevent AMI event documentation from being pulled in to the core.xml documentation file. Because of this, the `LOG_GROUP` documentation added in 89709e2 was not being properly extracted and was missing fom the resulting XML documentation file. This commit moves the `LOG_GROUP` documentation to a separate `logger.xml` file.	2 years ago
Sean Bright	8c3ebf9747	config.c: Log #exec include failures. ... If the script referenced by `#exec` does not exist, writes anything to stderr, or exits abnormally or with a non-zero exit status, we log that to Asterisk's error logging channel. Additionally, write out a warning if the script produces no output. Fixes #259	2 years ago
Sean Bright	ca931c9436	app.c: Allow ampersands in playback lists to be escaped. ... Any function or application that accepts a `&`-separated list of filenames can now include a literal `&` in a filename by wrapping the entire filename in single quotes, e.g.: ``` exten = _X.,n,Playback('https://example.com/sound.cgi?a=b&c=d'&hello-world) ``` Fixes #172 UpgradeNote: Ampersands in URLs passed to the `Playback()`, `Background()`, `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()` applications as filename arguments can now be escaped by single quoting the filename. Additionally, this is also possible when using the `CONFBRIDGE` dialplan function, or configuring various features in `confbridge.conf` and `queues.conf`.	2 years ago