6985 Commits (da8814524bc32a582d3cba00fee6d2d8fd78e108)

Author	SHA1	Message	Date
Jasper Hafkenscheid	da8814524b	channelstorage_cpp_map_name_id: Fix get_by_name_prefix prefix match ... Lower bound filter did not ensure prefix match. Resolves: #1730	8 months ago
Talha Asghar	0a76b99594	say.c: added language support for pashto and dari ... With this new feature, users who speak these languages can now benefit from the text-to-speech functionality provided by asterisk. This will make the platform more accessible and useful to a wider range of users, particularly those in regions where Pashto and Dari are spoken. This contribution will help to improve the overall usability and inclusivity of the asterisk platform. Fixes: #1724	8 months ago
Sean Bright	fb433fdba6	cli.c: Allow 'channel request hangup' to accept patterns. ... This extends 'channel request hangup' to accept multiple channel names, a POSIX Extended Regular Expression, a glob-like pattern, or a combination of all of them. UserNote: The 'channel request hangup' CLI command now accepts multiple channel names, POSIX Extended Regular Expressions, glob-like patterns, or a combination of all of them. See the CLI command 'core show help channel request hangup' for full details.	9 months ago
Sean Bright	7088b5c661	asterisk.c: Allow multi-byte characters on the Asterisk CLI. ... Versions of libedit that support Unicode expect that the EL_GETCFN (the function that does character I/O) will fill in a `wchar_t` with a character, which may be multi-byte. The built-in function that libedit provides, but does not expose with a public API, does properly handle multi-byte sequences. Due to the design of Asterisk's console processing loop, Asterisk provides its own implementation which does not handle multi-byte characters. Changing Asterisk to use libedit's built-in function would be ideal, but would also require changing some fundamental things about console processing which could be fairly disruptive. Instead, we bring in libedit's `read_char` implementation and modify it to suit our specific needs. Resolves: #60	9 months ago
sarangr7	c37d2cd0d7	main/dial.c: Set channel hangup cause on timeout in handle_timeout_trip ... When dial attempts timeout in the core dialing API, the channel's hangup cause was not being set before hanging up. Only the ast_dial_channel structure's internal cause field was updated, but the actual ast_channel hangup cause remained unset. This resulted in incorrect or missing hangup cause information being reported through CDRs, AMI events, and other mechanisms that read the channel's hangup cause when dial timeouts occurred via applications using the dialing API (FollowMe, Page, etc.). The fix adds proper channel locking and sets AST_CAUSE_NO_ANSWER on the channel before calling ast_hangup(), ensuring consistent hangup cause reporting across all interfaces. Resolves: #1660	9 months ago
Sean Bright	4dd0537a0f	http.c: Include remote address in URI handler message. ... Resolves: #1662	9 months ago
Joshua C. Colp	a12251ca88	pjsip: Move from threadpool to taskpool ... This change moves the PJSIP module from the threadpool API to the taskpool API. PJSIP-specific implementations for task usage have been removed and replaced with calls to the optimized taskpool implementations instead. The need for a pool of serializers has also been removed as taskpool inherently provides this. The default settings have also been changed to be more realistic for common usage. UpgradeNote: The threadpool_* options in pjsip.conf have now been deprecated though they continue to be read and used. They have been replaced with taskpool options that give greater control over the underlying taskpool used for PJSIP. An alembic upgrade script has been added to add these options to realtime as well.	9 months ago
George Joseph	3fe9377c90	endpoint.c: Plug a memory leak in ast_endpoint_shutdown(). ... Commit 26795be introduced a memory leak of ast_endpoint when ast_endpoint_shutdown() was called. The leak occurs only if a configuration change removes an endpoint and isn't related to call volume or the length of time asterisk has been running. An ao2_ref(-1) has been added to ast_endpoint_shutdown() to plug the leak. Resolves: #1635	10 months ago
Sean Bright	786156e3ba	Revert "func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()" ... This reverts commit 5177662990. For rationale, see #1621 and #1606	10 months ago
Mike Bradeen	5f2c9ffd0d	taskprocessors: Improve logging and add new cli options ... This change makes some small changes to improve log readability in addition to the following changes: Modified 'core show taskprocessors' to now show Low time and High time for task execution. New command 'core show taskprocessor name <taskprocessor-name>' to dump taskprocessor info and current queue. Addionally, a new test was added to demonstrate the 'show taskprocessor name' functionality: test execute category /main/taskprocessor/ name taskprocessor_cli_show Setting 'core set debug 3 taskprocessor.c' will now log pushed tasks. (Warning this is will cause extremely high levels of logging at even low traffic levels.) Resolves: #1566 UserNote: New CLI command has been added - core show taskprocessor name <taskprocessor-name>	10 months ago
Michal Hajek	3bd4852fa6	manager: fix double free of criteria variable when adding filter ... Signed-off-by: Michal Hajek <michal.hajek@daktela.com> Fixes: #1531	10 months ago
Sean Bright	abcf456715	abstract_jb.c: Remove redundant timer check per static analysis. ... While this check is technically unnecessary, it also was not harmful. The 2 other items mentioned in the linked issue are false positives and require no action. Resolves: #1417	10 months ago
phoneben	8dec33f963	channelstorage_cpp: Fix fallback return value in channelstorage callback ... callback returned the last iterated channel when no match existed, causing invalid channel references and potential double frees. Updated to correctly return NULL when there is no match. Resolves: #1609	10 months ago
George Joseph	724f4cb6e8	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.	10 months ago
Naveen Albert	d4ff1a10f4	pbx.c: Print new context count when reloading dialplan. ... When running "dialplan reload", the number of contexts reported is initially wrong, as it is the old context count. Running "dialplan reload" a second time returns the correct number of contexts that are loaded. This can confuse users into thinking that the reload didn't work successfully the first time. This counter is currently only incremented when iterating the old contexts prior to the context merge; at the very end, get the current number of elements in the context hash table and report that instead. This way, the count is correct immediately whenever a reload occurs. Resolves: #1599	10 months ago
Tinet-mucw	c1d8dead62	core_unreal.c: Use ast instead of p->chan to get the DIALSTATUS variable ... After p->chan = NULL, ast still points to the valid channel object, using ast safely accesses the channel's DIALSTATUS variable before it's fully destroyed Resolves: #1590	10 months ago
phoneben	0361a8e9d7	stasis: switch stasis show topics temporary container from list - RBtree ... switch stasis show topics temporary container from list to RB-tree minimizing lock time Resolves: #1585	11 months ago
Sean Bright	b64bd60df3	main: Explicitly mark case statement fallthrough as such. ... Resolves: #1442	11 months ago
George Joseph	687874c733	channelstorage: Allow storage driver read locking to be skipped. ... After PR #1498 added read locking to channelstorage_cpp_map_name_id, if ARI channels/externalMedia was called with a custom channel id AND the cpp_map_name_id channel storage backend is in use, a deadlock can occur when hanging up the channel. It's actually triggered in channel.c:__ast_channel_alloc_ap() when it gets a write lock on the channelstorage driver then subsequently does a lookup for channel uniqueid which now does a read lock. This is an invalid operation and causes the lock state to get "bad". When the channels try to hang up, a write lock is attempted again which hangs and causes the deadlock. Now instead of the cpp_map_name_id channelstorage driver "get" APIs automatically performing a read lock, they take a "lock" parameter which allows a caller who already has a write lock to indicate that the "get" API must not attempt its own lock. This prevents the state from getting mesed up. The ao2_legacy driver uses the ao2 container's recursive mutex so doesn't have this issue but since it also implements the common channelstorage API, it needed its "get" implementations updated to take the lock parameter. They just don't use it. Resolves: #1578	11 months ago
Joshua C. Colp	ff56cec9a1	Revert "pjsip: Move from threadpool to taskpool" ... This reverts commit bb6b76c2d8.	11 months ago
Tinet-mucw	7a891733b8	iostream.c: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions. ... The TCP three-way handshake completes, but if the server is under a TLS handshake attack, asterisk will get stuck at SSL_do_handshake(). In this case, a timeout mechanism should be set for the SSL/TLS handshake process to prevent indefinite waiting during the SSL handshake. Resolves: #1559	11 months ago
Joshua C. Colp	8d39faaf12	pjsip: Move from threadpool to taskpool ... This change moves the PJSIP module from the threadpool API to the taskpool API. PJSIP-specific implementations for task usage have been removed and replaced with calls to the optimized taskpool implementations instead. The need for a pool of serializers has also been removed as taskpool inherently provides this. The default settings have also been changed to be more realistic for common usage. UpgradeNote: The threadpool_* options in pjsip.conf have now been deprecated though they continue to be read and used. They have been replaced with taskpool options that give greater control over the underlying taskpool used for PJSIP. An alembic upgrade script has been added to add these options to realtime as well.	11 months ago
Joshua C. Colp	31530be124	devicestate: Don't publish redundant device state messages. ... When publishing device state check the local cache for the existing device state. If the new device state is unchanged from the prior one, don't bother publishing the update. This can reduce the work done by consumers of device state, such as hints and app_queue, by not publishing a message to them. These messages would most often occur with devices that are seeing numerous simultaneous channels. The underlying device state would remain as in use throughout, but an update would be published as channels are created and hung up.	11 months ago
George Joseph	72e9e4665e	chan_pjsip: Add technology-specific off-nominal hangup cause to events. ... Although the ISDN/Q.850/Q.931 hangup cause code is already part of the ARI and AMI hangup and channel destroyed events, it can be helpful to know what the actual channel technology code was if the call was unsuccessful. For PJSIP, it's the SIP response code. * A new "tech_hangupcause" field was added to the ast_channel structure along with ast_channel_tech_hangupcause() and ast_channel_tech_hangupcause_set() functions. It should only be set for off-nominal terminations. * chan_pjsip was modified to set the tech hangup cause in the chan_pjsip_hangup() and chan_pjsip_session_end() functions. This is a bit tricky because these two functions aren't always called in the same order. The channel that hangs up first will get chan_pjsip_session_end() called first which will trigger the core to call chan_pjsip_hangup() on itself, then call chan_pjsip_hangup() on the other channel. The other channel's chan_pjsip_session_end() function will get called last. Unfortunately, the other channel's HangupRequest events are sent before chan_pjsip has had a chance to set the tech hangupcause code so the HangupRequest events for that channel won't have the cause code set. The ChannelDestroyed and Hangup events however will have the code set for both channels. * A new "tech_cause" field was added to the ast_channel_snapshot_hangup structure. This is a public structure so a bit of refactoring was needed to preserve ABI compatibility. * The ARI ChannelHangupRequest and ChannelDestroyed events were modified to include the "tech_cause" parameter in the JSON for off-nominal terminations. The parameter is suppressed for nominal termination. * The AMI SoftHangupRequest, HangupRequest and Hangup events were modified to include the "TechCause" parameter for off-nominal terminations. Like their ARI counterparts, the parameter is suppressed for nominal termination. DeveloperNote: A "tech_cause" parameter has been added to the ChannelHangupRequest and ChannelDestroyed ARI event messages and a "TechCause" parameter has been added to the HangupRequest, SoftHangupRequest and Hangup AMI event messages. For chan_pjsip, these will be set to the last SIP response status code for off-nominally terminated calls. The parameter is suppressed for nominal termination.	11 months ago
Joshua C. Colp	133a2348ef	endpoints: Remove need for stasis subscription. ... When an endpoint is created in the core of Asterisk a subscription was previously created alongside it to monitor any channels being destroyed that were related to it. This was done by receiving all channel snapshot updates for every channel and only reacting when it was indicated that the channel was dead. This change removes this logic and instead provides an API call for directly removing a channel from an endpoint. This is called when channels are destroyed. This operation is fast, so blocking the calling thread for a short period of time doesn't have any noticeable impact.	12 months ago
George Joseph	ceb9007dfc	taskpool: Fix some references to threadpool that should be taskpool. ... Resolves: #1478	12 months ago
Naveen Albert	d2297a9a8d	core_unreal: Preserve ADSI capability when dialing Local channels. ... Dial() already preserves the ADSI capability by copying it to the new channel, but since Local channel pairs consist of two channels, we also need to copy the capability to the second channel. Resolves: #1517	12 months ago
Igor Goncharovsky	6c86dd3fd9	func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE() ... As soon as SIP call may end with several Reason headers, we want to make all of them available through the HAGUPCAUSE() function. This implementation uses the same ao2 hash for cause codes storage and adds a flag to make difference between last processed sip message and content of reason headers. UserNote: Added a new option to HANGUPCAUSE to access additional information about hangup reason. Reason headers from pjsip could be read using 'tech_extended' cause type.	12 months ago
Naveen Albert	bbc10f19fc	codec_builtin.c: Adjust some of the quality scores to reflect reality. ... Among the lower-quality voice codecs, some of the quality scores did not make sense relative to each other. For instance, quality-wise, G.729 > G.723 > PLC10. However, current scores do not uphold these relationships. Tweak the scores slightly to reflect more accurate relationships. Resolves: #1501	12 months ago
George Joseph	64adbfa1e7	channelstorage_cpp_map_name_id: Add read locking around retrievals. ... When we retrieve a channel from a C++ map, we actually get back a wrapper object that points to the channel then right after we retrieve it, we bump its reference count. There's a tiny chance however that between those two statements a delete and/or unref might happen which would cause the wrapper object or the channel itself to become invalid resulting in a SEGV. To avoid this we now perform a read lock on the driver around those statements. Resolves: #1491	12 months ago
Naveen Albert	8908cc5b96	dsp.c: Make minor fixes to debug log messages. ... Commit dc8e3eeaaf improved the debug log messages in dsp.c. This makes two minor corrections to it: * Properly guard an added log statement in a conditional. * Don't add one to the hit count if there was no hit (however, we do still want to do this for the case where this is one). Resolves: #1496	12 months ago
Naveen Albert	a51c7c14d6	config_options.c: Improve misleading warning. ... When running "config show help <module>", if no XML documentation exists for the specified module, "Module <module> not found." is returned, which is misleading if the module is loaded but simply has no XML documentation for its config. Improve the message to clarify that the module may simply have no config documentation. Resolves: #1489	12 months ago
Joshua C. Colp	b1e410c4b5	sorcery: Move from threadpool to taskpool. ... This change moves observer invocation from the use of a threadpool to a taskpool. The taskpool options have also been adjusted to ensure that at least one taskprocessor remains available at all times.	1 year ago
Sven Kube	aa3ff4ec60	stasis_channels.c: Make protocol_id optional to enable blind transfer via ari ... When handling SIP transfers via ARI, there is no protocol_id in case of a blind transfer. Resolves: #1467	1 year ago
Allan Nathanson	8340954c42	config.c: fix saving of deep/wide template configurations ... Follow-on to #244 and #960 regarding how the ast_config_XXX APIs handle template inheritance. ast_config_text_file_save2() incorrectly suppressed variables if they matched any ancestor template. This broke deep chains (dropping values based on distant parents) and wide inheritance (ignoring last-wins order across multiple parents). The function now inspects the full template hierarchy to find the nearest effective parent (last occurrence wins). Earlier inherited duplicates are collapsed, explicit overrides are kept unless they exactly match the parent, and PreserveEffectiveContext avoids writing redundant lines. Resolves: #1451	1 year ago
Bastian Triller	573c8faa09	Fix some doxygen, typos and whitespace	1 year ago
Sven Kube	30daba71bc	stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_create ... When handling SIP transfers via ARI, the `referred_by` field in `transfer_ari_state` may be null, since SIP REFER requests are not required to include a `Referred-By` header. Without this check, a null value caused the transfer to fail and triggered a NOTIFY with a 500 Internal Server Error.	1 year ago
Joshua C. Colp	e4aa5bbd22	taskpool: Update versions for taskpool stasis options.	1 year ago
Joshua C. Colp	0c49ba4478	taskpool: Add taskpool API, switch Stasis to using it. ... This change introduces a new API called taskpool. This is a pool of taskprocessors. It provides the following functionality: 1. Task pushing to a pool of taskprocessors 2. Synchronous tasks 3. Serializers for execution ordering of tasks 4. Growing/shrinking of number of taskprocessors in pool This functionality already exists through the combination of threadpool+taskprocessors but through investigating I determined that this carries substantial overhead for short to medium duration tasks. The threadpool uses a single queue of work, and for management of threads it involves additional tasks. I wrote taskpool to eliminate the extra overhead and management as much as possible. Instead of a single queue of work each taskprocessor has its own queue and at push time a selector chooses the taskprocessor to queue the task to. Each taskprocessor also has its own thread like normal. This spreads out the tasks immediately and reduces contention on shared resources. Using the included efficiency tests the number of tasks that can be executed per second in a taskpool is 6-12 times more than an equivalent threadpool+taskprocessor setup. Stasis has been moved over to using this new API as it is a heavy consumer of threadpool+taskprocessors and produces a lot of tasks. UpgradeNote: The threadpool_* options in stasis.conf have now been deprecated though they continue to be read and used. They have been replaced with taskpool options that give greater control over the underlying taskpool used for stasis. DeveloperNote: The taskpool API has been added for common usage of a pool of taskprocessors. It is suggested to use this API instead of the threadpool+taskprocessor approach.	1 year ago
Nathan Monfils	1657ccfc68	manager.c: Fix presencestate object leak ... ast_presence_state allocates subtype and message. We straightforwardly need to clean those up.	1 year ago
Sean Bright	d274905fa5	audiohook.c: Ensure correct AO2 reference is dereffed. ... Part of #1440.	1 year ago
Ben Ford	a85cbadf83	rtp_engine.c: Add exception for comfort noise payload. ... In a previous commit, a change was made to ast_rtp_codecs_payload_code_tx_sample_rate to check for differing sample rates. This ended up returning an invalid payload int for comfort noise. A check has been added that returns early if the payload is in fact supposed to be comfort noise. Fixes: #1340	1 year ago
Naveen Albert	d9cbadc51b	pbx_variables.c: Create real channel for "dialplan eval function". ... "dialplan eval function" has been using a dummy channel for function evaluation, much like many of the unit tests. However, sometimes, this can cause issues for functions that are not expecting dummy channels. As an example, ast_channel_tech(chan) is NULL on such channels, and ast_channel_tech(chan)->type consequently results in a NULL dereference. Normally, functions do not worry about this since channels executing dialplan aren't dummy channels. While some functions are better about checking for these sorts of edge cases, use a real channel with a dummy technology to make this CLI command inherently safe for any dialplan function that could be evaluated from the CLI. Resolves: #1434	1 year ago
Naveen Albert	171d5e996c	pbx_builtins: Allow custom tone for WaitExten. ... 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.	1 year ago
Alexei Gradinari	14ea4d3b39	sorcery: Prevent duplicate objects and ensure missing objects are created on update ... This patch resolves two issues in Sorcery objectset handling with multiple backends: 1. 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. 2. 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.	1 year ago
George Joseph	5d822d64ef	chan_websocket: Allow additional URI parameters to be added to the outgoing URI. ... * Added a new option to the WebSocket dial string to capture the additional URI parameters. * Added a new API ast_uri_verify_encoded() that verifies that a string either doesn't need URI encoding or that it has already been encoded. * Added a new API ast_websocket_client_add_uri_params() to add the params to the client websocket session. * Added XML documentation that will show up with `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.	1 year ago
Naveen Albert	dd9c71ed0b	dsp.c: Improve debug logging in tone_detect(). ... 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: * If we had a DSP hit, don't also emit the previous debug message that was always logged. It is duplicated by the hit message, so this can reduce the number of debug messages during detection by 50%. * Include the hit count and required number of hits in the message so on partial detections can be more easily troubleshot. * Use debug level 9 for hits instead of 10, so we can focus on hits without all the noise from the per-frame debug message. * 1-index the hit count in the debug messages. On the first hit, it currently logs '0', just as when we are not detecting anything, which can be confusing. Resolves: #1375	1 year ago
George Joseph	59c9aa00b0	xmldoc.c: Fix rendering of CLI output. ... 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. * Fixed the rendering of examples so they are indented properly and changed the color so they can be seen. * There is now a line break before each option. * Options are now printed on their own line with all option content indented below them. 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 <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. ``` 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.	1 year ago
Naveen Albert	1ddf63d83f	func_frame_drop: Add debug messages for dropped frames. ... Add debug messages in scenarios where frames that are usually processed are dropped or skipped. Resolves: #1371	1 year ago
Naveen Albert	a23c467c82	bridge.c: Obey BRIDGE_NOANSWER variable to skip answering channel. ... 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	1 year ago