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.
The tone used while waiting for digits in WaitExten can now be overridden by specifying an argument for the 'd' option.
The 'e' option for TONE_DETECT now allows detection to be disabled automatically once the desired number of matches have been fulfilled, which can help prevent race conditions in the dialplan, since TONE_DETECT does not need to be disabled after a hit.
Users relying on Sorcery multiple writable backends configurations (e.g., astdb + realtime) may now enable update_or_create_on_update_miss = yes in sorcery.conf to ensure missing objects are recreated after temporary backend failures. Default behavior remains unchanged unless explicitly enabled.
A new WebSocket channel driver option v has been added to the
Dial application that allows you to specify additional URI parameters on
outgoing connections. Run core show application Dial from the Asterisk CLI
to see how to use it.
ChanSpy and ExtenSpy can now be configured to not automatically answer the channel by using the 'N' option.
Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types in cel.conf will log media files and music on hold played to each channel. The STREAM_BEGIN event's extra field will contain a JSON with the file details (path, format and language), or the class name, in case of music on hold is played. The DTMF event's extra field will contain a JSON with the digit and the duration in milliseconds.
Options are now available in the menuselect "Resource Modules" category that allow you to enable the AES_192, AES_256 and AES_GCM cipher suites in res_srtp. Of course, libsrtp and OpenSSL must support them but modern versions do. Previously, the only way to enable them was to set the CFLAGS environment variable when running ./configure. The default setting is to disable them preserving existing behavior.
A new CDR option "canceldispositionenabled" has been added that when set to true, the NO ANSWER disposition will be split into two dispositions: CANCEL and NO ANSWER. The default value is 'no'
The httpauth field in CURLOPT now allows the authentication methods to be set.
A new channel driver "chan_websocket" is now available. It can exchange media over both inbound and outbound websockets and will both frame and re-time the media it receives. See http://s.asterisk.net/mow for more information. The ARI channels/externalMedia API now includes support for the
A new ARI endpoint is available at /channels/{channelId}/progress to indicate progress to a channel.
The 32-bit ast_options has no room left to accomodate new options and so has been converted to an ast_flags64 structure. All internal references to ast_options have been updated to use the 64-bit flag manipulation macros. External module references to the 32-bit ast_options should continue to work on little-endian systems because the least-significant bytes of a 64 bit integer will be in the same location as a 32-bit integer. Because that's not the case on big-endian systems, we've swapped the bytes in the flags manupulation macros on big-endian systems so external modules should still work however you are encouraged to test.
sorcery: Prevent duplicate objects and ensure missing objects are created on u..
pbx_lua.c: segfault when pass null data to term_color function
file.c: with "sounds_search_custom_dir = yes", search "custom" directory
Fix missing ast_test_flag64 in extconf.c
res_rtp_asterisk: Don't send RTP before DTLS has negotiated.
chan_websocket: Allow additional URI parameters to be added to the outgoing URI.
app_queue.c: Add new global 'log_unpause_on_reason_change'
res_musiconhold: Appropriately lock channel during start.
chan_websocket.c: Add DTMF messages
res_stasis_device_state: Fix delete ARI Devicestates after asterisk restart.
chan_pjsip.c: Change SSRC after media source change
res_pjsip_diversion: resolve race condition between Diversion header processin..
pbx_builtins: Allow custom tone for WaitExten.
res_musiconhold.c: Ensure we're always locked around music state access.
cel: Add STREAM_BEGIN, STREAM_END and DTMF event types.
bundled_pjproject: Avoid deadlock between transport and transaction
app_queue: fix comparison for announce-position-only-up
ARI: Add command to indicate progress to a channel
pbx.c: When the AST_SOFTHANGUP_ASYNCGOTO flag is set, pbx_extension_helper sho..
chan_websocket: Reset frame_queue_length to 0 after FLUSH_MEDIA
utils.h: Add rounding to float conversion to int.
Author: Joe Garlick Date: 2025-09-04
Added DTMF messages to the chan_websocket feature.
When a user presses DTMF during a call over chan_websocket it will send a message like: "DTMF_END digit:1"
Resolves: https://github.com/asterisk/asterisk-feature-requests/issues/70
Author: Igor Goncharovsky Date: 2025-09-02
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.
Author: Igor Goncharovsky Date: 2025-09-04
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
Author: Artem Umerov Date: 2025-08-29
Fix missing ast_test_flag64 after https://github.com/asterisk/asterisk/commit/43bf8a4ded7a65203b766b91eaf8331a600e9d8d
Author: Naveen Albert Date: 2025-08-25
Currently, the 'd' option will play dial tone while waiting for digits. Allow it to accept an argument for any tone from indications.conf.
Resolves: #1396
UserNote: The tone used while waiting for digits in WaitExten can now be overridden by specifying an argument for the 'd' option.
Author: Naveen Albert Date: 2025-08-28
One of the problems with TONE_DETECT as it was originally written is that if a tone is detected multiple times, it can trigger the redirect logic multiple times as well. For example, if we do an async goto in the dialplan after detecting a tone, because the detector is still active until explicitly disabled, if we detect the tone again, we will branch again and start executing that dialplan a second time. This is rarely ever desired behavior, and can happen if the detector is not removed quickly enough.
Add a new option, 'e', which automatically disables the detector once the desired number of matches have been heard. This eliminates the potential race condition where previously the detector would need to be disabled immediately, but doing so quickly enough was not guaranteed. This also allows match criteria to be retained longer if needed, so the detector does not need to be destroyed prematurely.
Resolves: #1390
UserNote: The 'e' option for TONE_DETECT now allows detection to be disabled automatically once the desired number of matches have been fulfilled, which can help prevent race conditions in the dialplan, since TONE_DETECT does not need to be disabled after a hit.
Author: Stuart Henderson Date: 2025-08-21
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
Author: Alexei Gradinari Date: 2025-07-07
This patch resolves two issues in Sorcery objectset handling with multiple backends:
Prevent duplicate objects: When an object exists in more than one backend (e.g., a contact in both 'astdb' and 'realtime'), the objectset previously returned multiple instances of the same logical object. This caused logic failures in components like the PJSIP registrar, where duplicate contact entries led to overcounting and incorrect deletions, when max_contacts=1 and remove_existing=yes.
This patch ensures only one instance of an object with a given key is added to the objectset, avoiding these duplicate-related side effects.
Ensure missing objects are created: When using multiple writable backends, a temporary backend failure can lead to objects missing permanently from that backend. Currently, .update() silently fails if the object is not present, and no .create() is attempted. This results in inconsistent state across backends (e.g. astdb vs. realtime).
This patch introduces a new global option in sorcery.conf: [general] update_or_create_on_update_miss = yes|no
Default: no (preserves existing behavior).
When enabled: if .update() fails with no data found, .create() is attempted in that backend. This ensures that objects missing due to temporary backend outages are re-synchronized once the backend is available again.
Added a new CLI command: sorcery show settings Displays global Sorcery settings, including the current value of update_or_create_on_update_miss.
Updated tests to validate both flag enabled/disabled behavior.
Fixes: #1289
UserNote: Users relying on Sorcery multiple writable backends configurations (e.g., astdb + realtime) may now enable update_or_create_on_update_miss = yes in sorcery.conf to ensure missing objects are recreated after temporary backend failures. Default behavior remains unchanged unless explicitly enabled.
Author: Naveen Albert Date: 2025-08-25
If Caller ID is disabled for an FXS port, then we should not send any Caller ID spill on the line, as we have no Caller ID information that we can/should be sending.
Resolves: #1394
Author: Naveen Albert Date: 2025-08-18
It is possible to modify the dialmode setting in the chan_dahdi/sig_analog private using the CHANNEL function, to modify it during calls. However, it was not being reset between calls, meaning that if, for example, tone dialing was disabled, it would never work again unless explicitly enabled.
This fixes the setting by pairing it with a "perm" version of the setting, as a few other features have, so that it can be reset to the permanent setting between calls. The documentation is also clarified to explain the interaction of this setting and the digitdetect setting more clearly.
Resolves: #1378
Author: George Joseph Date: 2025-08-13
core show application Dial
that shows how to use it.Resolves: #1352
UserNote: A new WebSocket channel driver option v has been added to the
Dial application that allows you to specify additional URI parameters on
outgoing connections. Run core show application Dial from the Asterisk CLI
to see how to use it.
Author: George Joseph Date: 2025-08-19
ast_websocket_read() receives data into a fixed 64K buffer then continually reallocates a final buffer that, after all continuation frames have been received, is the exact length of the data received and returns that to the caller. process_text_message() in chan_websocket was attempting to set a NULL terminator on the received payload assuming the payload buffer it received was the large 64K buffer. The assumption was incorrect so when it tried to set a NULL terminator on the payload, it could, depending on the state of the heap at the time, cause heap corruption.
process_text_message() now allocates its own payload_len + 1 sized buffer, copies the payload received from ast_websocket_read() into it then NULL terminates it prevent the possibility of the overrun and corruption.
Resolves: #1384
Author: Naveen Albert Date: 2025-08-18
Add an additional check to guard against the channel application being NULL.
Resolves: #1380
Author: Sven Kube Date: 2025-07-30
Adds an ARI command to send a progress indication to a channel.
DeveloperNote: A new ARI endpoint is available at /channels/{channelId}/progress to indicate progress to a channel.
Author: Naveen Albert Date: 2025-08-15
The debug logging during DSP processing has always been kind of overwhelming and annoying to troubleshoot. Simplify and improve the logging in a few ways to aid DSP debugging:
Resolves: #1375
Author: Jose Lopes Date: 2025-07-30
After an asterisk restart, the deletion of ARI Devicestates didn't return error, but the devicestate was not deleted. Found a typo on populate_cache function that created wrong cache for device states. This bug caused wrong assumption that devicestate didn't exist, since it was not in cache, so deletion didn't returned error.
Fixes: #1327
Author: Naveen Albert Date: 2025-08-13
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.
Author: George Joseph Date: 2025-08-14
If you do a core show application Dial, you'll see it's kind of a mess.
Indents are wrong is some places, examples are printed in black which makes
them invisible on most terminals, and the lack of line breaks in some cases
makes it hard to follow.
Example from Dial before fixes: ``` Example: Dial 555-1212 on first available channel in group 1, searching from highest to lowest
Example: Ringing FXS channel 4 with ring cadence 2
Example: Dial 555-1212 on channel 3 and require answer confirmation
...
O([mode]):
mode - With <mode> either not specified or set to '1', the originator
hanging up will cause the phone to ring back immediately.
With
p: This option enables screening mode. This is basically Privacy mode without memory. ```
After: ``` Example: Dial 555-1212 on first available channel in group 1, searching from highest to lowest
same => n,Dial(DAHDI/g1/5551212)
Example: Ringing FXS channel 4 with ring cadence 2
same => n,Dial(DAHDI/4r2)
Example: Dial 555-1212 on channel 3 and require answer confirmation
same => n,Dial(DAHDI/3c/5551212)
...
O([mode]):
mode - With <mode> either not specified or set to '1', the originator
hanging up will cause the phone to ring back immediately.
With <mode> set to '2', when the operator flashes the trunk, it will
ring their phone back.
Enables *operator services* mode. This option only works when bridging
a DAHDI channel to another DAHDI channel only. If specified on
non-DAHDI interfaces, it will be ignored. When the destination answers
(presumably an operator services station), the originator no longer has
control of their line. They may hang up, but the switch will not
release their line until the destination party (the operator) hangs up.
p:
This option enables screening mode. This is basically Privacy mode
without memory.
```
There are still things we can do to make this more readable but this is a start.
Author: Naveen Albert Date: 2025-08-14
Add debug messages in scenarios where frames that are usually processed are dropped or skipped.
Resolves: #1371
Author: Naveen Albert Date: 2025-08-14
curl_easy_setopt expects long types, so be explicit.
Resolves: #1369
Author: Naveen Albert Date: 2025-08-14
Handle allocation failure and simplify the allocation using asprintf.
Resolves: #1366
Author: Alexey Khabulyak Date: 2025-08-14
This can be reproduced under certain curcomstences. For example: call app.playback from lua with invalid data: app.playback({}). pbx_lua.c will try to get data for this playback using lua_tostring function. This function returs NULL for everything but strings and numbers. Then, it calls term_color with NULL data. term_color function can call(if we don't use vt100 compat term) ast_copy_string with NULL inbuf which cause segfault. bt example: ast_copy_string (size=8192, src=0x0, dst=0x7fe44b4be8b0) at /usr/src/asterisk/asterisk-20.11.0/include/asterisk/strings.h:412
Resolves: https://github.com/asterisk/asterisk/issues/1363
Author: Naveen Albert Date: 2025-08-14
If the BRIDGE_NOANSWER variable is set on a channel, it is not supposed to answer when another channel bridges to it using Bridge(), and this is checked when ast_bridge_call* is called. However, another path exists (bridge_exec -> ast_bridge_add_channel) where this variable was not checked and channels would be answered. We now check the variable there.
Resolves: #401 Resolves: #1364
Author: Ben Ford Date: 2025-08-04
There was no check in __rtp_sendto that prevented Asterisk from sending RTP before DTLS had finished negotiating. This patch adds logic to do so.
Fixes: #1260
Author: Alexey Khabulyak Date: 2025-08-04
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
Author: Mike Bradeen Date: 2025-08-07
Based on the firing order of the PJSIP call-backs on a redirect, it was possible for the Diversion header to not be included in the outgoing 181 response to the UAC and the INVITE to the UAS.
This change moves the Diversion header processing to an earlier PJSIP callback while also preventing the corresponding update that can cause a duplicate 181 response when processing the header at that time.
Resolves: #1349
Author: Allan Nathanson Date: 2025-08-10
With sounds_search_custom_dir = yes, we are supposed to search for sounds
in the AST_DATA_DIR/sounds/custom directory before searching the normal
directories. Unfortunately, a recent change
(https://github.com/asterisk/asterisk/pull/1172) had a typo resulting in
the "custom" directory not being searched. This change restores this
expected behavior.
Resolves: #1353
Author: Sperl Viktor Date: 2025-06-30
Fixes: #1280
UserNote: Enabling the tracking of the STREAM_BEGIN and the STREAM_END event types in cel.conf will log media files and music on hold played to each channel. The STREAM_BEGIN event's extra field will contain a JSON with the file details (path, format and language), or the class name, in case of music on hold is played. The DTMF event's extra field will contain a JSON with the digit and the duration in milliseconds.
Author: George Joseph Date: 2025-07-30
The fact that deleting an object from a map invalidates any iterator that happens to currently point to that object was overlooked in the initial implementation. Unfortunately, there's no way to detect that an iterator has been invalidated so the result was an occasional SEGV triggered by modules like app_chanspy that opens an iterator and can keep it open for a long period of time. The new implementation doesn't keep the underlying C++ iterator open across calls to ast_channel_iterator_next() and uses a read lock on the map to ensure that, even for the few microseconds we use the iterator, another thread can't delete a channel from under it. Even with this change, the iterators are still WAY faster than the ao2_legacy storage driver.
Full details about the new implementation are located in the comments for iterator_next() in channelstorage_cpp_map_name_id.cc.
Resolves: #1309
Author: George Joseph Date: 2025-08-05
UserNote: Options are now available in the menuselect "Resource Modules" category that allow you to enable the AES_192, AES_256 and AES_GCM cipher suites in res_srtp. Of course, libsrtp and OpenSSL must support them but modern versions do. Previously, the only way to enable them was to set the CFLAGS environment variable when running ./configure. The default setting is to disable them preserving existing behavior.
Author: zhou_jiajian Date: 2025-07-24
In the original implementation, both CANCEL and NO ANSWER states were consolidated under the NO ANSWER disposition. This patch introduces a separate CANCEL disposition, with an optional configuration switch to enable this new disposition.
Resolves: #1323
UserNote: A new CDR option "canceldispositionenabled" has been added that when set to true, the NO ANSWER disposition will be split into two dispositions: CANCEL and NO ANSWER. The default value is 'no'
Author: Naveen Albert Date: 2025-08-01
Currently the CURL function only supports Basic Authentication, the default auth method in libcurl. Add an option that also allows enabling digest authentication.
Resolves: #1332
UserNote: The httpauth field in CURLOPT now allows the authentication methods to be set.
Author: George Joseph Date: 2025-07-21
DeveloperNote: The 32-bit ast_options has no room left to accomodate new options and so has been converted to an ast_flags64 structure. All internal references to ast_options have been updated to use the 64-bit flag manipulation macros. External module references to the 32-bit ast_options should continue to work on little-endian systems because the least-significant bytes of a 64 bit integer will be in the same location as a 32-bit integer. Because that's not the case on big-endian systems, we've swapped the bytes in the flags manupulation macros on big-endian systems so external modules should still work however you are encouraged to test.
Author: Alexei Gradinari Date: 2025-07-15
This patch fixes an issue in the ODBC Realtime engine where Asterisk incorrectly falls back to the next configured backend when the current one returns SQL_NO_DATA (i.e., no record found). This is a logical error and performance risk in multi-backend configurations.
Solution: Introduced CONFIG_RT_NOT_FOUND ((void *)-1) as a special return marker. ODBC Realtime backend now return CONFIG_RT_NOT_FOUND when no data is found. Core engine stops iterating on this marker, avoiding unnecessary fallback.
Notes: Other Realtime backends (PostgreSQL, LDAP, etc.) can be updated similarly. This patch only covers ODBC.
Fixes: #1305
Author: Sven Kube Date: 2025-07-30
ast_ari_channels_create and ast_ari_channels_dial called the
ast_channel_get_by_name function with optional arguments. Since
8f1982c4d6, this function logs an error for empty channel names.
This commit adds checks for empty optional arguments that are used
to call ast_channel_get_by_name to prevent these error logs.
Author: Naveen Albert Date: 2025-07-28
The already-deprecated "password" option for the AGENT function was removed in commit d43b17a872e8227aa8a9905a21f90bd48f9d5348 for Asterisk 12, but the documentation for it wasn't removed then.
Resolves: #1321
Author: Tinet-mucw Date: 2025-07-22
Under certain circumstances the context/extens/prio are set in the ast_async_goto, for example action Redirect. In the situation that action Redirect is broken by pbx_extension_helper this info is changed. This will cause the current dialplan location to be executed twice. In other words, the Redirect action does not take effect.
Resolves: #1315
Author: Sperl Viktor Date: 2025-07-22
Fixes: #1317
Author: Naveen Albert Date: 2025-07-16
Currently, the ast_tls_cert script is hardcoded to produce certificates with a validity of 365 days, which is not generally desirable for self- signed certificates. Make this parameter configurable.
Resolves: #1307
Author: George Joseph Date: 2025-07-17
The CDR tenantid was being set in cdr_object_alloc from the channel->base
snapshot. Since this happens at channel creation before the dialplan is even
reached, calls to CHANNEL(tenantid)=<something> in the dialplan were being
ignored. Instead we now take tenantid from party_a when
cdr_object_create_public_records() is called which is after the call has
ended and all channel snapshots rebuilt. This is exactly how accountcode
and amaflags, which can also be set in tha dialplpan, are handled.
Resolves: #1259
Author: George Joseph Date: 2025-07-16
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
Author: Naveen Albert Date: 2025-07-14
Previously, we were only using # (ST) as a terminator, and not handling A (STP), B (ST2P), or C (ST3P), which erroneously led to it being treated as part of the dialed number. Parse any of these as the start digit.
Resolves: #1301
Author: kodokaii Date: 2025-07-03
In the WebSocket channel driver, the FLUSH_MEDIA command clears all frames from the queue but does not reset the frame_queue_length counter.
As a result, the driver incorrectly thinks the queue is full after flushing, which prevents new multimedia frames from being sent, especially after multiple flush commands.
This fix sets frame_queue_length to 0 after flushing, ensuring the queue state is consistent with its actual content.
Fixes: #1304
Author: Martin Tomec Date: 2025-06-25
When the RTP media source changes, such as after a blind transfer, the new source introduces a discontinuous timestamp. According to RFC 3550, Section 5.1, an RTP stream's timestamp for a given SSRC must increment monotonically and linearly. To comply with the standard and avoid a large timestamp jump on the existing SSRC, a new SSRC is generated for the new media stream. This change resolves known interoperability issues with certain SBCs (like Sonus/Ribbon) that stop forwarding media when they detect such a timestamp violation. This code uses the existing implementation from chan_sip.
Resolves: #927
Author: George Joseph Date: 2025-04-28
Created chan_websocket which can exchange media over both inbound and outbound websockets which the driver will frame and time. See http://s.asterisk.net/mow for more information.
res_http_websocket: Made defines for max message size public and converted a few nuisance verbose messages to debugs.
main/channel.c: Changed an obsolete nuisance error to a debug.
ARI channels: Updated externalMedia to include chan_websocket as a supported transport.
UserNote: A new channel driver "chan_websocket" is now available. It can exchange media over both inbound and outbound websockets and will both frame and re-time the media it receives. See http://s.asterisk.net/mow for more information.
UserNote: The ARI channels/externalMedia API now includes support for the WebSocket transport provided by chan_websocket.
Author: Stanislav Abramenkov Date: 2025-07-01
Backport patch from upstream * Avoid deadlock between transport and transaction https://github.com/pjsip/pjproject/commit/edde06f261ac
Issue described in https://github.com/pjsip/pjproject/issues/4442
Author: mkmer Date: 2025-03-23
Quote from an audio engineer NR9V:
There is a minor issue of a small amount of crossover distortion though as a result of ast_slinear_saturated_multiply_float() not rounding the float. This could result in some quiet but potentially audible distortion artifacts in lower volume parts of the signal. If you have for example a sign wave function with a max amplitude of just a few samples, all samples between -1 and 1 will be truncated to zero, resulting in the waveform no longer being a sine wave and in harmonic distortion.
Resolves: #1176
Author: Tinet-mucw Date: 2025-06-18
Under certain circumstances the context/extens/prio are set in the ast_async_goto, for example action Redirect. In the situation that action Redirect is broken by GotoIf this info is changed. that will causes confusion in dialplan execution.
Resolves: #1273
Author: Sean Bright Date: 2025-04-08
Author: Sean Bright Date: 2025-04-08
Author: Jaco Kroon Date: 2024-12-19
This relates to #829
This doesn't sully solve the Ops issue, but it solves the specific crash there. Further PRs to follow.
In the specific crash the generator was still under construction when moh was being stopped, which then proceeded to close the stream whilst it was still in use.
Signed-off-by: Jaco Kroon jaco@uls.co.za