asterisk

Commit Graph

Author	SHA1	Message	Date
Naveen Albert	359320c952	app_dial: Include channel name in warnings during wait_for_answer. Include the channel name in warnings during wait_for_answer to make them more useful and allow problematic channels to be easily identified. Resolves: #1802	6 months ago
Naveen Albert	03bc82ab23	build: Fix unused-but-set-variable warnings with gcc 16. Fix or remove a few variables that were being set but not actually used anywhere, causing warnings with gcc 16. Resolves: #1783	6 months ago
Joshua C. Colp	6965d570af	build: Fix GCC discarded-qualifiers const errors. GCC 15.2.1 pays attention to the discarding of the const qualifier when strchr, strrchr, memchr, or memrchr are now used. This change fixes numerous errors with this throughout the tree. The fixes can be broken down into the following: 1. The return value should be considered const. 2. The value passed to strchr or strrchr can be cast as it is expected and allowed to be modified. 3. The pointer passed to strchr or strrchr is not meant to be modified and so the contents must be duplicated. 4. It was declared const and never should have been.	7 months ago
Naveen Albert	90f0f75407	app_dial, utils.h: Avoid old style declaration and discarded qualifier. * app_dial: Use const char* for fixed strings. * utils.h: inline should come before return type. Resolves: #1755	7 months ago
serfreeman1337	1d9c7fc4bb	app_queue: Add 'prio' setting to the 'force_longest_waiting_caller' option This adds a 'prio' setting to ensure that call priority is respected across multiple queues. Using 'yes' could cause high-priority callers to be skipped if a caller in another queue had a longer wait time, regardless of priority. Resolves: #1637 UserNote: The 'force_longest_waiting_caller' option now supports a 'prio' setting. When set to 'prio', calls are offered by priority first, then by wait time.	7 months ago
phoneben	48309d8be7	app_queue: Queue Timing Parity with Dial() and Accurate Wait Metrics app_queue: Set Dial-compatible timing variables Extends Queue() to set Dial-compatible timing variables (ANSWEREDTIME, DIALEDTIME) and introduces a precise QUEUEWAIT metric calculated at agent connect time, with proper initialization to prevent stale or misleading values.	7 months ago
phoneben	9496fe61b1	app_queue: Fix rN raise_penalty ignoring min_penalty in calc_metric QUEUE_RAISE_PENALTY=rN was not respected during member selection. calc_metric() raised penalties below QUEUE_MIN_PENALTY, allowing excluded members to be selected. This change makes calc_metric() honor raise_respect_min, keeping behavior consistent with queue empty checks and expected rN semantics UserNote: Fixes an issue where QUEUE_RAISE_PENALTY=rN could raise a member’s penalty below QUEUE_MIN_PENALTY during member selection. This could allow members intended to be excluded to be selected. The queue now consistently respects the minimum penalty when raising penalties, aligning member selection behavior with queue empty checks and documented rN semantics.	7 months ago
serfreeman1337	1e517b7835	app_queue: Only compare calls at 1st position across queues when forcing longest waiting caller. This prevents a situation where a call joining at 1st position to a queue with calls leads to a state where no callers are considered the longest waiting, causing queues to stop offering calls. Resolves: #1691	7 months ago
George Joseph	640ea61f21	app_amd: Remove errant space in documentation for totalAnalysisTime.	8 months ago
Tinet-mucw	f53a8f15e8	app_mixmonitor.c: Fix crash in mixmonitor_ds_remove_and_free when datastore is NULL The datastore may be NULL, so a null pointer check needs to be added. Resolves: #1673	8 months ago
Sean Bright	5d6ba4d3e0	app_queue.c: Update docs to correct QueueMemberPause event name.	9 months ago
Sean Bright	8b0d7243ce	app_stream_echo.c: Check that stream is non-NULL before dereferencing. Also re-order and rename the arguments of `stream_echo_write_error` to match those of `ast_write_stream` for consistency. Resolves: #1427	9 months ago
George Joseph	bb6bf8a3f9	ccss: Add option to ccss.conf to globally disable it. The Call Completion Supplementary Service feature is rarely used but many of it's functions are called by app_dial and channel.c "just in case". These functions lock and unlock the channel just to see if CCSS is enabled on it, which it isn't 99.99% of the time. UserNote: A new "enabled" parameter has been added to ccss.conf. It defaults to "yes" to preserve backwards compatibility but CCSS is rarely used so setting "enabled = no" in the "general" section can save some unneeded channel locking operations and log message spam. Disabling ccss will also prevent the func_callcompletion and chan_dahdi modules from loading. DeveloperNote: A new API ast_is_cc_enabled() has been added. It should be used to ensure that CCSS is enabled before making any other ast_cc_* calls.	9 months ago
George Joseph	6156cdcb7d	app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE. UpgradeNote: In an effort to reduce log spam, two normal progress "pickup attempted" log messages from app_directed_pickup have been changed from NOTICE to VERBOSE(3). This puts them on par with other normal dialplan progress messages.	9 months ago
Naveen Albert	951ead13d8	app_reload: Fix Reload() without arguments. Calling Reload() without any arguments is supposed to reload everything (equivalent to a 'core reload'), but actually does nothing. This is because it was calling ast_module_reload with an empty string, and the argument needs to explicitly be NULL. Resolves: #1597	10 months ago
Naveen Albert	66f337ac51	app_disa: Avoid use of removed ResetCDR() option. Commit `a46d5f9b76` removed the deprecated 'e' option to ResetCDR; this now causes DISA() to emit a warning if attempting to call ResetCDR() with the deprecated option (in all cases except when the no answer option is provided). Rewrite the code to do this the current way. Resolves: #1592	10 months ago
Daouda Taha	2e699dd95c	app_mixmonitor: Add 's' (skip) option to delay recording. The 's' (skip) option delays MixMonitor recording until the specified number of seconds (can be fractional) have elapsed since MixMonitor was invoked. No audio is written to the recording file during this time. If the call ends before this period, no audio will be saved. This is useful for avoiding early audio such as announcements, ringback tones, or other non-essential sounds. UserNote: This change introduces a new 's(<seconds>)' (skip) option to the MixMonitor application. Example: MixMonitor(${UNIQUEID}.wav,s(3)) This skips recording for the first 3 seconds before writing audio to the file. Existing MixMonitor behavior remains unchanged when the 's' option is not used.	10 months ago
Sean Bright	4a7c94a9cc	app_dtmfstore: Avoid a potential buffer overflow. Prefer snprintf() so we can readily detect if our output was truncated. Resolves: #1421	10 months ago
Kristian F. Høgh	092147cc8c	app_queue.c: Only announce to head caller if announce_to_first_user Only make announcements to head caller if announce_to_first_user is true Fixes: #1568 UserNote: When announce_to_first_user is false, no announcements are played to the head caller	10 months ago
Ben Ford	d5e88e7ed9	app_queue.c: Fix error in Queue parameter documentation. When macro was removed in Asterisk 21, the parameter documentation in code was not updated to reflect the correct numerization for gosub. It still stated that it was the seventh parameter, but got shifted to the sixth due to the removal of macro. This has been updated to correctly reflect the parameter order, and a note has been added to the XML that states this was done after the initial commit. Fixes: #1534 UpgradeNote: As part of Asterisk 21, macros were removed from Asterisk. This resulted in argument order changing for the Queue dialplan application since the macro argument was removed. Upgrade notice was missed when this was done, so this upgrade note has been added to provide a record of such and a notice to users who may have not upgraded yet.	11 months ago
Joshua C. Colp	6944a7d2d7	app_queue: Allow stasis message filtering to work. The app_queue module subscribes on a per-dialed agent basis to both the bridge all and channel all topics to keep apprised of things going on involving them. This subscription has associated state that must be cleaned up when the subscription ends. This was done by setting a default router callback that only had logic to handle the case where the subscription ends. By using the default router callback all filtering for the subscription was disabled, causing unrelated messages to get published and handled by it. This change makes it so that an explicit route is added for the message type used for the message indicating the subscription has ended and removes the default router callback. This allows message filtering to occur on publishing reducing the messages to app_queue to only those it is interested in.	11 months ago
Naveen Albert	f647f9dad9	app_sf: Add post-digit timer option to ReceiveSF. Add a sorely needed option to set a timeout between digits, rather than for receiving the entire number. This is needed if the number of digits being sent is unknown by the receiver in advance. Previously, we had to wait for the entire timer to expire. Resolves: #1493 UserNote: The 't' option for ReceiveSF now allows for a timer since the last digit received, in addition to the number-wide timeout.	11 months ago
Naveen Albert	d8bc249f2e	app_dial: Allow fractional seconds for dial timeouts. Even though Dial() internally uses milliseconds for its dial timeouts, this capability has been mostly obscured from users as the argument is only parsed as an integer, thus forcing the use of whole seconds for timeouts. Parse it as a decimal instead so that timeouts can now truly have millisecond precision. Resolves: #1487 UserNote: The answer and progress dial timeouts now have millisecond precision, instead of having to be whole numbers.	11 months ago
Bastian Triller	f68cfae069	Fix some doxygen, typos and whitespace	12 months ago
phoneben	edcd4d0411	app_queue: Add NULL pointer checks in app_queue Add NULL check for word_list before calling word_in_list() Add NULL checks for channel snapshots from ast_multi_channel_blob_get_channel() Resolves: #1425	12 months ago
Sean Bright	c88febe23a	app_externalivr: Prevent out-of-bounds read during argument processing. Resolves: #1422	12 months ago
Naveen Albert	33ec85c3fd	app_adsiprog: Fix possible NULL dereference. get_token can return NULL, but process_token uses this result without checking for NULL; as elsewhere, check for a NULL result to avoid possible NULL dereference. Resolves: #1419	1 year ago
Igor Goncharovsky	63868245c0	app_queue.c: Add new global 'log_unpause_on_reason_change' In many asterisk-based systems, the pause reason is used to separate pauses by type,and logically, changing the reason defines two intervals that should be accounted for separately. The introduction of a new option allows me to separate the intervals of operator inactivity in the log by the event of unpausing. UserNote: Add new global option 'log_unpause_on_reason_change' that is default disabled. When enabled cause addition of UNPAUSE event on every re-PAUSE with reason changed.	1 year ago
Igor Goncharovsky	a8afc93ebe	app_waitforsilence.c: Use milliseconds to calculate timeout time The functions WaitForNoise() and WaitForSilence() use the time() functions to calculate elapsed time, which causes the timer to fire on a whole second boundary, and the actual function execution time to fire the timer may be 1 second less than expected. This fix replaces time() with ast_tvnow(). Fixes: #1401	1 year ago
Stuart Henderson	ac2948dc89	app_queue: fix comparison for announce-position-only-up Numerically comparing that the current queue position is less than last_pos_said can only be done after at least one announcement has been made, otherwise last_pos_said is at the default (0). Fixes: #1386	1 year ago
Naveen Albert	1cdead8a06	app_chanspy: Add option to not automatically answer channel. Add an option for ChanSpy and ExtenSpy to not answer the channel automatically. Most applications that auto-answer by default already have an option to disable this behavior if unwanted. Resolves: #1358 UserNote: ChanSpy and ExtenSpy can now be configured to not automatically answer the channel by using the 'N' option.	1 year ago
Alexey Khabulyak	09a3c8df70	app_dial.c: Moved channel lock to prevent deadlock It's reproducible with pbx_lua, not regular dialplan. deadlock description: 1. asterisk locks a channel 2. calls function onedigit_goto 3. calls ast_goto_if_exists funciton 4. checks ast_exists_extension -> pbx_extension_helper 5. pbx_extension_helper calls pbx_find_extension 6. Then asterisk starts autoservice in a new thread 7. autoservice run tries to lock the channel again Because our channel is locked already, autoservice can't lock. Autoservice can't lock -> autoservice stop is waiting forever. onedigit_goto waits for autoservice stop. Resolves: https://github.com/asterisk/asterisk/issues/1335	1 year ago
Naveen Albert	1e0205fb68	app_agent_pool: Remove documentation for removed option. The already-deprecated "password" option for the AGENT function was removed in commit `d43b17a872` for Asterisk 12, but the documentation for it wasn't removed then. Resolves: #1321	1 year ago
George Joseph	a4da66e584	app_mixmonitor: Update the documentation concerning the "D" option. When using the "D" option to output interleaved audio, the file extension must be ".raw". That info wasn't being properly rendered in the markdown and HTML on the documentation site. The XML was updated to move the note in the option section to a warning in the description. Resolves: #1269	1 year ago
Naveen Albert	8ee49a67f5	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.	1 year ago
Itzanh	d3a9bd29da	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.	1 year ago
phoneben	a31b68c37a	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.	1 year ago
George Joseph	65199303b9	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	1 year ago
Naveen Albert	907b1083ad	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.	1 year ago
Naveen Albert	2cacaa5464	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	1 year ago
Naveen Albert	73c3fb1391	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	1 year ago
Luz Paz	af49027ccc	docs: Fix typos in apps/ Found via codespell	1 year ago
phoneben	487584987f	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.	1 year ago
Sean Bright	c11d28b13e	app_confbridge: Prevent crash when publishing channel-less event. Resolves: #1190	1 year ago
Florent CHAUVEAU	c2444b188a	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,,#...).	1 year ago
Norm Harrison	a6ef41e4ca	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>	1 year ago
Ben Ford	091171c0eb	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.	1 year ago
Sean Bright	f042fb2153	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.	2 years ago
Jeremy Lainé	85538b6d71	docs: Fix minor typo in MixMonitor AMI action The `Options` argument was erroneously documented as lowercase `options`.	2 years ago
Sean Bright	67e89b3e77	docs: Indent <since> tags. Also updates the 'since' of applications/functions that existed before XML documentation was introduced (1.6.2.0).	2 years ago

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5340 Commits (82cf46fefbb7a6cbce9cf25684fde637628f285d)