5298 Commits (87b58f398ba699cd53f7b03338c9630e2f23d75f)

Author	SHA1	Message	Date
Naveen Albert	eb6808e580	app_record: Add RECORDING_INFO function. ... Add a function that can be used to retrieve info about a previous recording, such as its duration. This is being added as a function to avoid possibly trampling on dialplan variables, and could be extended to provide other information in the future. Resolves: #548 UserNote: The RECORDING_INFO function can now be used to retrieve the duration of a recording.	3 weeks ago
Itzanh	2928b876d8	app_sms.c: Fix sending and receiving SMS messages in protocol 2 ... This fixes bugs in SMS messaging to SMS-capable analog phones that prevented app_sms.c from talking to phones using SMS protocol 2. - Fix MORX message reception (from phone to Asterisk) in SMS protocol 2 - Fix MTTX message transmission (from Asterisk to phone) in SMS protocol 2 One of the bugs caused messages to have random characters and junk appended at the end up to the character limit. Another bug prevented Asterisk from sending messages from Asterisk to the phone at all. A final bug caused the transmission from Asterisk to the phone to take a long time because app_sms.c did not hang up after correctly sending the message, causing the phone to have to time out and hang up in order to complete the message transmission. This was tested with a Linksys PAP2T and with a GrandStream HT814, sending and receiving messages with Telefónica DOMO Mensajes phones from Telefónica Spain. I had to play with both the network jitter buffer and the dB gain to get it to work. One of my phones required the gain to be set to +3dB for it to work, while another required it to be set to +6dB. Only MORX and MTTX were tested, I did not test sending and receiving messages to a TelCo SMSC.	3 weeks ago
phoneben	76e22b0670	app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penalties only for members within min/max range ... This update adds support for a new QUEUE_RAISE_PENALTY format: rN When QUEUE_RAISE_PENALTY is set to rN (e.g., r4), only members whose current penalty is greater than or equal to the defined min_penalty and less than or equal to max_penalty will have their penalty raised to N. Members with penalties outside the min/max range remain unchanged. Example behaviors: QUEUE_RAISE_PENALTY=4 → Raise all members with penalty < 4 (existing behavior) QUEUE_RAISE_PENALTY=r4 → Raise only members with penalty in [min_penalty, max_penalty] to 4 Implementation details: Adds parsing logic to detect the r prefix and sets the raise_respect_min flag Modifies the raise logic to skip members outside the defined penalty range when the flag is active UserNote: This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective penalty raises only for members whose current penalty is within the [min_penalty, max_penalty] range. Members with lower or higher penalties are unaffected. This behavior is backward-compatible with existing queue rule configurations.	3 weeks ago
George Joseph	f82c7b2694	ARI Outbound Websockets ... Asterisk can now establish websocket sessions _to_ your ARI applications as well as accepting websocket sessions _from_ them. Full details: http://s.asterisk.net/ari-outbound-ws Code change summary: * Added an ast_vector_string_join() function, * Added ApplicationRegistered and ApplicationUnregistered ARI events. * Converted res/ari/config.c to use sorcery to process ari.conf. * Added the "outbound-websocket" ARI config object. * Refactored res/ari/ari_websockets.c to handle outbound websockets. * Refactored res/ari/cli.c for the sorcery changeover. * Updated res/res_stasis.c for the sorcery changeover. * Updated apps/app_stasis.c to allow initiating per-call outbound websockets. * Added CLI commands to manage ARI websockets. * Added the new "outbound-websocket" object to ari.conf.sample. * Moved the ARI XML documentation out of res_ari.c into res/ari/ari_doc.xml UserNote: Asterisk can now establish websocket sessions _to_ your ARI applications as well as accepting websocket sessions _from_ them. Full details: http://s.asterisk.net/ari-outbound-ws	3 weeks ago
Naveen Albert	bfbbe28344	sig_analog: Add Call Waiting Deluxe support. ... Adds support for Call Waiting Deluxe options to enhance the current call waiting feature. As part of this change, a mechanism is also added that allows a channel driver to queue an audio file for Dial() to play, which is necessary for the announcement function. ASTERISK-30373 #close Resolves: #271 UserNote: Call Waiting Deluxe can now be enabled for FXS channels by enabling its corresponding option.	3 weeks ago
Naveen Albert	6da909283d	app_sms: Ignore false positive vectorization warning. ... Ignore gcc warning about writing 32 bytes into a region of size 6, since we check that we don't go out of bounds for each byte. This is due to a vectorization bug in gcc 15, stemming from gcc commit 68326d5d1a593dc0bf098c03aac25916168bc5a9. Resolves: #1088	3 weeks ago
Naveen Albert	aa55410b58	app_meetme: Remove inaccurate removal version from xmldocs. ... app_meetme is deprecated but wasn't removed as planned in 21, so remove the inaccurate removal version. Resolves: #1224 (cherry picked from commit 9070302126)	5 months ago
Luz Paz	ec6b2091eb	docs: Fix typos in apps/ ... Found via codespell (cherry picked from commit b877d20d75)	5 months ago
phoneben	f42db69ed0	Add log-caller-id-name option to log Caller ID Name in queue log ... Add log-caller-id-name option to log Caller ID Name in queue log This patch introduces a new global configuration option, log-caller-id-name, to queues.conf to control whether the Caller ID name is logged when a call enters a queue. When log-caller-id-name=yes, the Caller ID name is logged as parameter 4 in the queue log, provided it’s allowed by the existing log_restricted_caller_id rules. If log-caller-id-name=no (the default), the Caller ID name is omitted from the logs. Fixes: #1091 UserNote: This patch adds a global configuration option, log-caller-id-name, to queues.conf to control whether the Caller ID name is logged as parameter 4 when a call enters a queue. When log-caller-id-name=yes, the Caller ID name is included in the queue log, Any '|' characters in the caller ID name will be replaced with '_'. (provided it’s allowed by the existing log_restricted_caller_id rules). When log-caller-id-name=no (the default), the Caller ID name is omitted. (cherry picked from commit f843f24be4)	5 months ago
Sean Bright	a45f1e0afa	app_confbridge: Prevent crash when publishing channel-less event. ... Resolves: #1190 (cherry picked from commit efeb102507)	5 months ago
Florent CHAUVEAU	d032df902b	audiosocket: added support for DTMF frames ... Updated the AudioSocket protocol to allow sending DTMF frames. AST_FRAME_DTMF frames are now forwarded to the server, in addition to AST_FRAME_AUDIO frames. A new payload type AST_AUDIOSOCKET_KIND_DTMF with value 0x03 was added to the protocol. The payload is a 1-byte ascii representing the DTMF digit (0-9,*,#...). UserNote: The AudioSocket protocol now forwards DTMF frames with payload type 0x03. The payload is a 1-byte ascii representing the DTMF digit (0-9,*,#...). (cherry picked from commit c44e7e85ec)	5 months ago
Norm Harrison	428c14d1f1	audiosocket: fix timeout, fix dialplan app exit, server address in logs ... - Correct wait timeout logic in the dialplan application. - Include server address in log messages for better traceability. - Allow dialplan app to exit gracefully on hangup messages and socket closure. - Optimize I/O by reducing redundant read()/write() operations. Co-authored-by: Florent CHAUVEAU <florentch@pm.me> (cherry picked from commit 218c64c4eb)	5 months ago
Ben Ford	a7c5da35a6	documentation: Update Gosub, Goto, and add new documentationtype. ... Gosub and Goto were not displaying their syntax correctly on the docs site. This change adds a new way to specify an optional context, an optional extension, and a required priority that the xml stylesheet can parse without having to know which optional parameters come in which order. In Asterisk, it looks like this: parameter name="context" documentationtype="dialplan_context" parameter name="extension" documentationtype="dialplan_extension" parameter name="priority" documentationtype="dialplan_priority" required="true" The stylesheet will ignore the context and extension parameters, but for priority, it will automatically inject the following: [[context,]extension,]priority This is the correct oder for applications such as Gosub and Goto. (cherry picked from commit 11684bc0d3)	7 months ago
Sean Bright	1008578d58	docs: AMI documentation fixes. ... Most of this patch is adding missing PJSIP-related event documentation, but the one functional change was adding a sorcery to-string handler for endpoint's `redirect_method` which was not showing up in the AMI event details or `pjsip show endpoint <endpoint>` output. The rest of the changes are summarized below: * app_agent_pool.c: Typo fix Epoche -> Epoch. * stasis_bridges.c: Add missing AttendedTransfer properties. * stasis_channels.c: Add missing AgentLogoff properties. * pjsip_manager.xml: - Add missing AorList properties. - Add missing AorDetail properties. - Add missing ContactList properties. - Add missing ContactStatusDetail properties. - Add missing EventDetail properties. - Add missing AuthList properties. - Add missing AuthDetail properties. - Add missing TransportDetail properties. - Add missing EndpointList properties. - Add missing IdentifyDetail properties. * res_pjsip_registrar.c: Add missing InboundRegistrationDetail documentation. * res_pjsip_pubsub.c: - Add missing ResourceListDetail documentation. - Add missing InboundSubscriptionDetail documentation. - Add missing OutboundSubscriptionDetail documentation. * res_pjsip_outbound_registration.c: Add missing OutboundRegistrationDetail documentation. (cherry picked from commit e86f937e3a)	7 months ago
Jeremy Lainé	1b55bcaad0	docs: Fix minor typo in MixMonitor AMI action ... The `Options` argument was erroneously documented as lowercase `options`. (cherry picked from commit 575545b49c)	7 months ago
Sean Bright	ee0a50e749	docs: Indent <since> tags. ... Also updates the 'since' of applications/functions that existed before XML documentation was introduced (1.6.2.0). (cherry picked from commit 174006fcaa)	7 months ago
George Joseph	e6fe538d81	docs: Add version information to application and function XML elements ... * Do a git blame on the embedded XML application or function element. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The applications and functions defined in the following files were done by hand because the XML was extracted from the C source file relatively recently. * channels/pjsip/dialplan_functions_doc.xml * main/logger_doc.xml * main/manager_doc.xml * res/res_geolocation/geoloc_doc.xml * res/res_stir_shaken/stir_shaken_doc.xml (cherry picked from commit 54d67711f8)	8 months ago
George Joseph	3fc90e7810	docs: Add version information to manager event instance XML elements ... * Do a git blame on the embedded XML managerEvent elements. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The events defined in res/res_pjsip/pjsip_manager.xml were done by hand because the XML was extracted from the C source file relatively recently. Two bugs were fixed along the way... * The get_documentation awk script was exiting after it processed the first DOCUMENTATION block it found in a file. We have at least 1 source file with multiple DOCUMENTATION blocks so only the first one in them was being processed. The awk script was changed to continue searching rather than exiting after the first block. * Fixing the awk script revealed an issue in logger.c where the third DOCUMENTATION block contained a XML fragment that consisted only of a managerEventInstance element that wasn't wrapped in a managerEvent element. Since logger_doc.xml already existed, the remaining fragments in logger.c were moved to it and properly organized. (cherry picked from commit 2897d87a99)	8 months ago
George Joseph	ded24e3375	docs: Add version information to configObject and configOption XML elements ... Most of the configObjects and configOptions that are implemented with ACO or Sorcery now have `<since>/<version>` elements added. There are probably some that the script I used didn't catch. The version tags were determined by the following... * Do a git blame on the API call that created the object or option. * From the commit hash, grab the summary line. * Do a `git log --grep <summary>` to find the cherry-pick commits in all branches that match. * Do a `git patch-id` to ensure the commits are all related and didn't get a false match on the summary. * Do a `git tag --contains <commit>` to find the tags that contain each commit. * Weed out all tags not <major>.<minor>.0. * Sort and discard any <major>.0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the API was last touched. configObjects and configOptions elements implemented with the base ast_config APIs were just not possible to find due to the non-deterministic way they are accessed. Also note that if the API call was on modified after it was added, the version will be the one it was last modified in. Final note: The configObject and configOption elements were introduced in 12.0.0 so options created before then may not have any XML documentation. (cherry picked from commit f70670841b)	8 months ago
Sean Bright	1197ebc6cd	manager: Add `<since>` tags for all AMI actions. ... (cherry picked from commit cede8a3e15)	8 months ago
Sperl Viktor	effb2f3af3	app_queue: indicate the paused state of a dynamically added member in queue_log. ... Fixes: #1021 (cherry picked from commit 38ef522fd2)	8 months ago
Sperl Viktor	63f366850a	app_queue: allow dynamically adding a queue member in paused state. ... Fixes: #1007 UserNote: use the p option of AddQueueMember() for paused member state. Optionally, use the r(reason) option to specify a custom reason for the pause. (cherry picked from commit 1689aaa576)	8 months ago
Ben Ford	00ceb1b69e	app_mixmonitor: Add 'D' option for dual-channel audio. ... Adds the 'D' option to app_mixmonitor that interleaves the input and output frames of the channel being recorded in the monitor output frame. This allows for two streams in the recording: the transmitted audio and the received audio. The 't' and 'r' options are compatible with this. Fixes: #945 UserNote: The MixMonitor application now has a new 'D' option which interleaves the recorded audio in the output frames. This allows for stereo recording output with one channel being the transmitted audio and the other being the received audio. The 't' and 't' options are compatible with this. (cherry picked from commit 6c2cbac384)	11 months ago
Naveen Albert	e66f69d45e	app_dial: Fix progress timeout calculation with no answer timeout. ... If to_answer is -1, simply comparing to see if the progress timeout is smaller than the answer timeout to prefer it will fail. Add an additional check that chooses the progress timeout if there is no answer timeout (or as before, if the progress timeout is smaller). Resolves: #821 (cherry picked from commit 6a6f962acb)	11 months ago
Naveen Albert	edb99e6cc6	app_dial: Fix progress timeout. ... Under some circumstances, the progress timeout feature added in commit 320c98eec8 does not work as expected, such as if there is no media flowing. Adjust the waitfor call to explicitly use the progress timeout if it would be reached sooner than the answer timeout to ensure we handle the timers properly. Resolves: #821 (cherry picked from commit b4dadd8ea5)	11 months ago
Sean Bright	06d8aac6fd	res_agi.c: Ensure SIGCHLD handler functions are properly balanced. ... Calls to `ast_replace_sigchld()` and `ast_unreplace_sigchld()` must be balanced to ensure that we can capture the exit status of child processes when we need to. This extends to functions that call `ast_replace_sigchld()` and `ast_unreplace_sigchld()` such as `ast_safe_fork()` and `ast_safe_fork_cleanup()`. The primary change here is ensuring that we do not call `ast_safe_fork_cleanup()` in `res_agi.c` if we have not previously called `ast_safe_fork()`. Additionally we reinforce some of the documentation and add an assertion to, ideally, catch this sooner were this to happen again. Fixes #922 (cherry picked from commit 750333ae03)	11 months ago
Naveen Albert	b4f28e30e2	app_voicemail: Fix ill-formatted pager emails with custom subject. ... Add missing end-of-headers newline to pager emails with custom subjects, since this was missing from this code path. Resolves: #902 (cherry picked from commit eb624a5aa2)	11 months ago
George Joseph	84d983e8f5	Fix application references to Background ... The app is actually named "BackGround" but several references in XML documentation were spelled "Background" with the lower case "g". This was causing documentation links to return "not found" messages. (cherry picked from commit 240f5c3a71)	11 months ago
Tinet-mucw	169637277f	app_chanspy.c: resolving the issue writing frame to whisper audiohook. ... ChanSpy(${channel}, qEoSw): because flags set o, ast_audiohook_set_frame_feed_direction(audiohook, AST_AUDIOHOOK_DIRECTION_READ); this will effect whisper audiohook and spy audiohook, this makes writing frame to whisper audiohook impossible. So add function start_whispering to starting whisper audiohook. Resolves: #876 (cherry picked from commit f8f5cec2c1)	1 year ago
Jean-Denis Girard	28d05dfae6	app_voicemail: Fix sql insert mismatch caused by cherry-pick ... When commit e8c9cb80 was cherry-picked in from master, the fact that the 20 and 18 branches still had the old "macrocontext" column wasn't taken into account so the number of named parameters didn't match the number of '?' placeholders. They do now. We also now use ast_asprintf to create the full mailbox query SQL statement instead of trying to calculate the proper length ourselves. Resolves: #831 (cherry picked from commit 59db5366ed)	1 year ago
Tinet-mucw	81f2d13e20	app_chanspy.c: resolving the issue with audiohook direction read ... ChanSpy(${channel}, qEoS): When chanspy spy the direction read, reading frame is often failed when reading direction read audiohook. because chanspy only read audiohook direction read; write_factory_ms will greater than 100ms soon, then ast_slinfactory_flush will being called, then direction read will fail. Resolves: #861 (cherry picked from commit 6e47295800)	1 year ago
George Joseph	0905dfdc1c	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. (cherry picked from commit e8c9cb8021)	1 year ago
Alexei Gradinari	c3f78b36a0	app_queue: Add option to not log Restricted Caller ID to queue_log ... Add a queue option log-restricted-caller-id to strip the Caller ID when storing the ENTERQUEUE event in the queue log if the Caller ID is restricted. Resolves: #765 UpgradeNote: Add a new column to the queues table: queue_log_option_log_restricted ENUM('0','1','off','on','false','true','no','yes') to control whether the Restricted Caller ID will be stored in the queue log. UserNote: Add a Queue option log-restricted-caller-id to control whether the Restricted Caller ID will be stored in the queue log. If log-restricted-caller-id=no then the Caller ID will be stripped if the Caller ID is restricted. (cherry picked from commit 41d4db99cf)	1 year ago
Sean Bright	c78d46fb56	app_queue.c: Properly handle invalid strategies from realtime. ... The existing code sets the queue strategy to `ringall` but it is then immediately overwritten with an invalid one. Fixes #707 (cherry picked from commit 9de1bc7204)	1 year ago
George Joseph	b5865aef32	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. (cherry picked from commit 3fb9d89586)	1 year ago
Naveen Albert	40fd161575	app_dial: Add dial time for progress/ringing. ... Add a timeout option to control the amount of time to wait if no early media is received before giving up. This allows aborting early if the destination is not being responsive. Resolves: #588 UserNote: The timeout argument to Dial now allows specifying the maximum amount of time to dial if early media is not received. (cherry picked from commit e4adc962ca)	2 years ago
Naveen Albert	5ade71ec9a	app_voicemail: Properly reinitialize config after unit tests. ... Most app_voicemail unit tests were not properly cleaning up after themselves after running. This led to test mailboxes lingering around in the system. It also meant that if any unit tests in app_voicemail that create mailboxes were executed and the module was not unloaded/loaded again prior to running the test_voicemail_vm_info unit test, Asterisk would segfault due to an attempt to copy a NULL string. The load_config test did actually have logic to reinitialize the config after the test. However, this did not work in practice since load_config() would not reload the config since voicemail.conf had not changed during the test; thus, additional logic has been added to ensure that voicemail.conf is truly reloaded, after any unit tests which modify the users list. This prevents the SEGV due to invalid mailboxes lingering around, and also ensures that the system state is restored to what it was prior to the tests running. Resolves: #629 (cherry picked from commit 082966bdad)	2 years ago
Shaaah	395bcab787	app_queue.c : fix "queue add member" usage string ... Fixing bracket placement in the "queue add member" cli usage string. (cherry picked from commit c6ff7a6a51)	2 years ago
Naveen Albert	3469e515de	app_voicemail: Allow preventing mark messages as urgent. ... This adds an option to allow preventing callers from leaving messages marked as 'urgent'. Resolves: #619 UserNote: The leaveurgent mailbox option can now be used to control whether callers may leave messages marked as 'Urgent'. (cherry picked from commit 3c42b141d7)	2 years ago
George Joseph	6df5fbee65	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 (cherry picked from commit a433ed0d5a)	2 years ago
cmaj	db99ad892b	app_speech_utils.c: Allow partial speech results. ... Adds 'p' option to SpeechBackground() application. With this option, when the app timeout is reached, whatever the backend speech engine collected will be returned as if it were the final, full result. (This works for engines that make partial results.) Resolves: #572 UserNote: The SpeechBackground dialplan application now supports a 'p' option that will return partial results from speech engines that provide them when a timeout occurs. (cherry picked from commit 63b5a03538)	2 years ago
Mike Bradeen	ec69680801	app_chanspy: Add 'D' option for dual-channel audio ... Adds the 'D' option to app chanspy that causes the input and output frames of the spied channel to be interleaved in the spy output frame. This allows the input and output of the spied channel to be decoded separately by the receiver. If the 'o' option is also set, the 'D' option is ignored as the audio being spied is inherently one direction. Fixes: #569 UserNote: The ChanSpy application now accepts the 'D' option which will interleave the spied audio within the outgoing frames. The purpose of this is to allow the audio to be read as a Dual channel stream with separate incoming and outgoing audio. Setting both the 'o' option and the 'D' option and results in the 'D' option being ignored. (cherry picked from commit 0d17ee6930)	2 years ago
Naveen Albert	e2cb63120f	app_if: Fix next priority calculation. ... Commit fa3922a4d2 fixed a branching issue but "overshoots" when calculating the next priority. This fixes that; accompanying test suite tests have also been extended. Resolves: #560 (cherry picked from commit 64f9df5e59)	2 years ago
Sean Bright	61c78fc48b	app_confbridge: Don't emit warnings on valid configurations. ... The numeric bridge profile options `internal_sample_rate` and `maximum_sample_rate` are documented to accept the special values `auto` and `none`, respectively. While these values currently work, they also emit warnings when used which could be confusing for users. In passing, also ensure that we only accept the documented range of sample rate values between 8000 and 192000. Fixes #546 (cherry picked from commit f541f8d8c4)	2 years ago
Naveen Albert	d7948f5425	app_if: Fix faulty EndIf branching. ... This fixes faulty branching logic for the EndIf application. Instead of computing the next priority, which should be done for false conditionals or ExitIf, we should simply advance to the next priority. Resolves: #341 (cherry picked from commit 83a0cb51e5)	2 years ago
Naveen Albert	91127a618f	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 (cherry picked from commit 8f5581b0d0)	2 years ago
Maximilian Fridrich	8f9200d106	app_dial: Add option "j" to preserve initial stream topology of caller ... Resolves: #462 UserNote: The option "j" is now available for the Dial application which uses the initial stream topology of the caller to create the outgoing channels. (cherry picked from commit dcf58ee88f)	2 years ago
Sean Bright	06d86c41af	app_voicemail.c: Completely resequence mailbox folders. ... Resequencing is a process that occurs when we open a voicemail folder and discover that there are gaps between messages (e.g. `msg0000.txt` is missing but `msg0001.txt` exists). Resequencing involves shifting the existing messages down so we end up with a sequential list of messages. Currently, this process stops after reaching a threshold based on the message limit (`maxmsg`) configured on the current folder. However, if `maxmsg` is lowered when a voicemail folder contains more than `maxmsg + 10` messages, resequencing will not run completely leaving the mailbox in an inconsistent state. We now resequence up to the maximum number of messages permitted by `app_voicemail` (currently hard-coded at 9999 messages). Fixes #86 (cherry picked from commit 6556a92393)	2 years ago
Sean Bright	f19b74ad31	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`. (cherry picked from commit f8212d4594)	2 years ago
Matthew Fredrickson	eac9ad69a8	app_followme.c: Grab reference on nativeformats before using it ... Fixes a crash due to a lack of proper reference on the nativeformats object before passing it into ast_request(). Also found potentially similar use case bugs in app_chanisavail.c, bridge.c, and bridge_basic.c Fixes: #388 (cherry picked from commit 275f7911b5)	2 years ago