5351 Commits (8cd58fd39d7871b16cac3fec72a142fa984b4254)

Author	SHA1	Message	Date
Pengpeng Hou	59b74276f2	app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers ... The protocol 1 unpack helpers trusted externally controlled lengths and wrote them directly into fixed-size buffers in sms_t. Clamp the address, header, and body copies to the destination array sizes so malformed messages cannot overwrite adjacent state. Resolves: #GHSA-q9fr-m7g8-6ph5	2 months ago
Mike Bradeen	6144b6bcb7	app_mixmonitor: Fix duplex recording for non 8K codecs ... The native sampling of duplex recording is set to match the raw 8K output format. If one or more of the streams being recorded is above 8K, the frame size coming into the mixmonitor is too large and needs to be translated to 8K before being mixed into the stereo frame to avoid garbled and mistimed audio Fixes: #1779	2 months ago
UpBeta	e6699a9298	app_record: Fix hangup handling during beep playback ... When a hangup occurs while app_record is playing the initial beep, the application does not detect the hangup and continues running until the maxduration timeout expires. Replace the manual ast_streamfile() + ast_waitstream() sequence with ast_stream_and_wait(), which properly detects hangup and returns non-zero, allowing the application to exit immediately with RECORD_STATUS set to HANGUP. Resolves: #1950	3 months ago
smtcbn	209f8f012e	app_queue: fix double increment of member->calls with shared_lastcall ... Under high concurrency, update_queue() may be invoked multiple times for the same call, causing member->calls and queue-level counters to be incremented more than once. The existing starttime check is not atomic and allows concurrent execution paths to pass. Treat member->starttime as a single-use token and consume it via CAS to ensure the call is counted exactly once. This also prevents incorrect call distribution when using strategies such as fewestcalls. Observed as a regression after upgrading to 20.17. Resolves: #1736	3 months ago
Naveen Albert	7988d1138c	app_dial: Properly handle callee hangup while sending digits. ... If we are sending digits (either DTMF, MF, or SF) to the called channel after receiving progress or a wink, and the callee hangs up before we have finished sending it digits, there are several problems that can ensue: * If the callee hung up without answering, the calling channel would hang up and not continue in the dialplan. * If the callee *did* answer before hanging up, the answer was never passed through to the caller, since this gets "eaten" by the various digit streaming functions and is never processed by app_dial. This is generally an edge case that occurs due to some kind of signaling failure, but to better handle this: * Set to_answer to 0 to prevent hangup on the exit path, just like other parts of wait_for_answer. * Better document this usage of to_answer. * If the channel did answer while it was receiving digits, manually answer the calling channel before we abort. The call would not continue in the dialplan anyways (either before or after this fix), but technically the call was answered, so the CDRs should probably reflect that, and this mirrors the behavior of calls which normally do not continue. Resolves: #1915 UserNote: If a called channel sends progress or wink and the caller begins sending digits but the callee answers and then hangs up before digit sending can finish, the call is now answered before being disconnected. If the callee hangs up without answering, the call now continues in the dialplan.	4 months ago
phoneben	cef63b761f	app_queue: Fix raise_respect_min lost in copy_rules() breaking rN queue rules ... app_queue: Fix raise_respect_min not copied in copy_rules() causing rN rules to be ignored. `copy_rules()` never copied `raise_respect_min` into the per-call rule list, so the flag was always 0 when a timed penaltychange rule fired, making `rN` behave like plain `N` and raising members below `min_penalty` that should have been excluded. Also fixes `update_qe_rule()` not propagating the flag from `qe->pr` to `qe`, and dropping the `r` prefix when saving back to `QUEUE_RAISE_PENALTY`. Resolves: #1901	4 months ago
phoneben	9e5f769e80	app_voicemail_odbc: fix msgnum race and crash on failed STORE ... app_voicemail_odbc: fix msgnum race and crash on failed STORE Two concurrent callers leaving voicemail to the same mailbox could be assigned the same msgnum because ast_unlock_path() was called before STORE(), allowing a second thread to read the same LAST_MSG_INDEX() before the first INSERT committed. The losing thread got a duplicate key error, but execution continued into notify_new_message() -> RETRIEVE() because the STORE() return value was not checked. RETRIEVE() then fetched the winning thread's DB row, mmap'd its blob size against the locally truncated file, and crashed with SIGBUS. Hold the path lock through STORE() and bail out on failure. Fixes: #1653	4 months ago
Daniel Donoghue	4413a02d0c	stasis_broadcast: Add optional ARI broadcast with first-claim-wins ... Adds two optional modules: res_stasis_broadcast.so: Infrastructure for broadcasting a single incoming channel to multiple ARI applications with atomic first-claim-wins semantics. app_stasis_broadcast.so: Provides the StasisBroadcast() dialplan application which invokes the broadcast infrastructure. Both modules are self-contained; if neither is loaded there is zero runtime impact. Loading them does not alter existing Stasis or ARI behavior unless explicitly used. Key Features (only active when modules are loaded): Fisher-Yates shuffled broadcast dispatch for fair claim races Atomic claim operations using mutex + condition variable signaling Configurable broadcast timeouts Safe regex application filtering with validation to mitigate ReDoS risk Thread-safe channel variable snapshotting (channel locked during reads) Late-claim safety: broadcast context kept alive until after the Stasis session ends so concurrent claimants always receive 409 Conflict rather than 404 Not Found Memory safety via RAII_VAR, ast_json_ref/unref, and ao2 reference counting Components Added: res/res_stasis_broadcast.c: Core broadcast + claim logic apps/app_stasis_broadcast.c: StasisBroadcast() dialplan application include/asterisk/stasis_app_broadcast.h: Public API header res/ari/resource_events.c: Integrates POST /ari/events/claim endpoint rest-api/api-docs/events.json: New CallBroadcast and CallClaimed events Implementation Notes: Broadcast contexts reside in an ao2 hash container keyed by channel id. Each context holds atomic claim state, winner application name, timeout metadata, and a condition variable for waiters. Broadcast contexts are kept alive until after stasis_app_exec() returns so that concurrent claimants racing against the timeout always receive 409 Conflict. Broadcast dispatch calls stasis_app_send() directly for each matching application in shuffled order. Regex filters are validated with bounded length, group depth, quantified group count, and alternation limits to reduce pathological backtracking. Timeout calculation uses timespec arithmetic with overflow-safe millisecond remainder handling. Event JSON follows existing Stasis/ARI conventions; references are managed correctly to avoid leaks or double frees. Optional Nature / Impact: No changes to existing APIs, events, or applications when absent. Clean fallback: systems ignoring the modules behave identically to prior versions. Development was assisted by Claude (Anthropic). All generated code has been reviewed, tested, and is understood by the author. UserNote: New optional modules res_stasis_broadcast.so and app_stasis_broadcast.so enable broadcasting an incoming channel to multiple ARI applications. The first application to successfully claim (via POST /ari/events/claim) wins channel control. StasisBroadcast() dialplan application initiates broadcasts. CallBroadcast and CallClaimed events notify applications. When modules are not loaded, behavior is unchanged. DeveloperNote: New public APIs in stasis_app_broadcast.h: stasis_app_broadcast_channel(), stasis_app_claim_channel(), stasis_app_broadcast_winner(), and stasis_app_broadcast_wait(). New ARI event types (CallBroadcast, CallClaimed) added to events.json. All code is isolated; no existing ABI modified.	4 months ago
Naveen Albert	83a215ccee	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	b533d7f035	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	1decab6def	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.	6 months ago
Naveen Albert	46a8521b00	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	d6edc71a69	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	43b69949ec	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	80d7446cbd	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	a741676a9d	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	881a399139	app_amd: Remove errant space in documentation for totalAnalysisTime.	7 months ago
Tinet-mucw	f91f7d580d	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	3ce5308ac9	app_queue.c: Update docs to correct QueueMemberPause event name.	9 months ago
Sean Bright	9647a4f022	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	ed8d5c6430	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	23b5de4923	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	e38a5b7c87	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	9 months ago
Naveen Albert	ba9d9ec64a	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	9 months ago
Daouda Taha	7d104d69a9	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	61b2538e9b	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	1a594fcb21	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	a1eb178410	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.	10 months ago
Joshua C. Colp	19b0636d3e	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	341cb16378	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	a6b15bfb68	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	a0f1e460fe	Fix some doxygen, typos and whitespace	11 months ago
phoneben	af67321134	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	11 months ago
Sean Bright	7b96fff129	app_externalivr: Prevent out-of-bounds read during argument processing. ... Resolves: #1422	11 months ago
Naveen Albert	4aad564b93	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	12 months ago
Igor Goncharovsky	f77e76bab1	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.	12 months ago
Igor Goncharovsky	39c677e0df	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	12 months ago
Stuart Henderson	11e3567200	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	12 months ago
Naveen Albert	bdf2e2f84a	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	b0421fc87c	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	aca74d100c	app_queue: Remove redundant/deprecated function. ... QUEUE_MEMBER_COUNT has been deprecated since at least 1.6, for fully duplicating functionality available in the QUEUE_MEMBER function; remove it now. Resolves: #1341 UpgradeNote: The deprecated QUEUE_MEMBER_COUNT function has been removed; use QUEUE_MEMBER(<queue>,logged) instead.	1 year ago
Naveen Albert	4a3c9ea574	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
Naveen Albert	741ad37086	app_voicemail: Remove deprecated options. ... Remove the deprecated maxmessage and minmessage options, which have been superseded by maxsecs and minsecs since 1.6. Also remove the deprecated 'cz' language option (deprecated since 1.8.) Resolves: #1298 UpgradeNote: The deprecated maxmessage and minmessage options have been removed; use maxsecs and minsecs instead. The deprecated 'cz' language has also been removed; use 'cs' instead.	1 year ago
George Joseph	1f52ae3582	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	415daae95f	users.conf: Remove deprecated users.conf integration. ... users.conf was deprecated in Asterisk 21 and is now being removed for Asterisk 23, in accordance with the Asterisk deprecation policy. This consists of: * Removing integration with app_directory, app_voicemail, chan_dahdi, chan_iax2, and AMI. * users.conf was also partially used for res_phoneprov, and this remaining functionality is consolidated to a separate phoneprov_users.conf, used only by res_phoneprov. Resolves: #1292 UpgradeNote: users.conf has been removed and all channel drivers must be configured using their specific configuration files. The functionality previously in users.conf for res_phoneprov is now in phoneprov_users.conf.	1 year ago
Naveen Albert	2198126231	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	dbe7c0c5b3	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	c753fe44d1	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	c873f2ae7e	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	2ced79259a	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