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${ noResults }
650 Commits (master)
Author | SHA1 | Message | Date |
---|---|---|---|
|
e3ad4cea4e |
docs: Fix various typos in main/
Found via `codespell -q 3 -S "./CREDITS" -L abd,asent,atleast,childrens,contentn,crypted,dne,durationm,exten,inout,leapyear,nd,oclock,offsetp,ot,parm,parms,requestor,ser,slanguage,slin,thirdparty,varn,varns,ues` |
2 months ago |
|
82bef2ed90 |
docs: Indent <since> tags.
Also updates the 'since' of applications/functions that existed before XML documentation was introduced (1.6.2.0). |
3 months ago |
|
198b01c536 |
docs: Add version information to application and function XML elements
* Do a git blame on the embedded XML application or function element. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The applications and functions defined in the following files were done by hand because the XML was extracted from the C source file relatively recently. * channels/pjsip/dialplan_functions_doc.xml * main/logger_doc.xml * main/manager_doc.xml * res/res_geolocation/geoloc_doc.xml * res/res_stir_shaken/stir_shaken_doc.xml |
3 months ago |
|
77d5550d14 |
docs: Add version information to manager event instance XML elements
* Do a git blame on the embedded XML managerEvent elements. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The events defined in res/res_pjsip/pjsip_manager.xml were done by hand because the XML was extracted from the C source file relatively recently. Two bugs were fixed along the way... * The get_documentation awk script was exiting after it processed the first DOCUMENTATION block it found in a file. We have at least 1 source file with multiple DOCUMENTATION blocks so only the first one in them was being processed. The awk script was changed to continue searching rather than exiting after the first block. * Fixing the awk script revealed an issue in logger.c where the third DOCUMENTATION block contained a XML fragment that consisted only of a managerEventInstance element that wasn't wrapped in a managerEvent element. Since logger_doc.xml already existed, the remaining fragments in logger.c were moved to it and properly organized. |
3 months ago |
|
468601fd13 |
manager: Add `<since>` tags for all AMI actions.
|
3 months ago |
|
4280d7cd77 |
pbx.c: expand fields width of "core show hints"
The current width for "extension" is 20 and "device state id" is 20, which is too small. The "extension" field contains "ext"@"context", so 20 characters is not enough. The "device state id" field, for example for Queue pause state contains Queue:"queue_name"_pause_PSJIP/"endpoint", so the 20 characters is not enough. Increase the width of "extension" field to 30 characters and the width of the "device state id" field to 60 characters. Resolves: #770 UserNote: The fields width of "core show hints" were increased. The width of "extension" field to 30 characters and the width of the "device state id" field to 60 characters. |
10 months ago |
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6871d1cdfc |
Reduce startup/shutdown verbose logging
When started with a verbose level of 3, asterisk can emit over 1500 verbose message that serve no real purpose other than to fill up logs. When asterisk shuts down, it emits another 1100 that are of even less use. Since the testsuite runs asterisk with a verbose level of 3, and asterisk starts and stops for every one of the 700+ tests, the number of log messages is staggering. Besides taking up resources, it also makes it hard to debug failing tests. This commit changes the log level for those verbose messages to 5 instead of 3 which reduces the number of log messages to only a handful. Of course, NOTICE, WARNING and ERROR message are unaffected. There's also one other minor change... ast_context_remove_extension_callerid2() logs a DEBUG message instead of an ERROR if the extension you're deleting doesn't exist. The pjsip_config_wizard calls that function to clean up the config and has been triggering that annoying error message for years. Resolves: #582 |
1 year ago |
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e899a02465 |
pbx.c: Fix gcc 12 compiler warning.
Resolves: #277 |
2 years ago |
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e8f548c155 |
app_macro: Remove deprecated module.
For most modules that interacted with app_macro, this change is limited to no longer looking for the current context from the macrocontext when set. Additionally, the following modules are impacted: app_dial - no longer supports M^ connected/redirecting macro app_minivm - samples written using macro will no longer work. The sample needs a re-write app_queue - can no longer a macro on the called party's channel. Use gosub which is currently supported ccss - no callback macro, gosub only app_voicemail - no macro support channel - remove macrocontext and priority, no connected line or redirection macro options options - stdexten is deprecated to gosub as the default and only pbx - removed macrolock pbx_dundi - no longer look for macro snmp - removed macro context, exten, and priority ASTERISK-30304 Change-Id: I830daab293117179b8d61bd4df0d971a1b3d07f6 |
2 years ago |
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d25bf55de5 |
pbx.c: Simplify ast_context memory management.
Allocate all of the ast_context's character data in the structure's flexible array member and eliminate the clunky fake_context. This will simplify future changes to ast_context. Change-Id: I98357de75d8ac2b3c4c9f201223632e6901021ea |
3 years ago |
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4aa541683b |
GCC12: Fixes for 16+
Most issues were in stringfields and had to do with comparing a pointer to an constant/interned string with NULL. Since the string was a constant, a pointer to it could never be NULL so the comparison was always "true". gcc now complains about that. There were also a few issues where determining if there was enough space for a memcpy or s(n)printf which were fixed by defining some of the involved variables as "volatile". There were also a few other miscellaneous fixes. ASTERISK-30044 Change-Id: Ia081ca1bcfb329df6487c4660aaf1944309eb570 |
3 years ago |
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6ddb0ec939 |
func_evalexten: Extension evaluation function.
This adds the EVAL_EXTEN function, which may be used to retrieve the variable-substituted data at any extension. ASTERISK-29486 Change-Id: Iad81019689674c9f4ac77d235f5d7234adbb1432 |
3 years ago |
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a7cf3979ec |
pbx.c: Warn if there are too many includes in a context.
The PBX core uses the stack when it comes to includes, which means that a context can only contain strictly fewer than AST_PBX_MAX_STACK includes. If this is exceeded, then warnings will be emitted for each number of includes beyond this if searching for an extension in the including context, and if the extension's inclusion is beyond the stack size, it will simply not be found. To address this, we now check if there are too many includes in a context when the dialplan is reloaded so that if there is an issue, the user is aware of at "compile time" as opposed to "run time" only. Secondly, more details are printed out when this message is encountered so it's clear what has happened. ASTERISK-26719 Change-Id: Ia3700452e75a7af3391b3e82ee69f06a669f8958 |
3 years ago |
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23a4a12420 |
pbx: Add variable substitution API for extensions
Currently, variable substitution involving dialplan extensions is quite clunky since it entails obtaining the current dialplan location, backing it up, storing the desired variables for substitution on the channel, performing substitution, then restoring the original location. In addition to being clunky, things could also go wrong if an async goto were to occur and change the dialplan location during a substitution. Fundamentally, there's no reason it needs to be done this way, so new API is added to allow for directly passing in the dialplan location for the purposes of variable substitution so we don't need to mess with the channel information anymore. Existing API is not changed. ASTERISK-29745 #close Change-Id: I23273bf27fa0efb64a606eebf9aa8e2f41a065e4 |
3 years ago |
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9440f6ec58 |
main: Fix for Doxygen.
ASTERISK-29763 Change-Id: Ib8359e3590a9109eb04a5376559d040e5e21867e |
3 years ago |
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cc025026b7 |
progdocs: Fix for Doxygen, the hidden parts.
ASTERISK-29779 Change-Id: If338163488498f65fa7248b60e80299c0a928e4b |
3 years ago |
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f382775241 |
main: Spelling fixes
Correct typos of the following word families: analysis nuisance converting although transaction desctitle acquire update evaluate thousand this dissolved management integrity reconstructed decrement further on irrelevant currently constancy anyway unconstrained featuregroups right larger evaluated encumbered languages digits authoritative framing blindxfer tolerate traverser exclamation perform permissions rearrangement performing processing declension happily duplicate compound hundred returns elicit allocate actually paths inheritance atxferdropcall earlier synchronization multiplier acknowledge across against thousands joyous manipulators guaranteed emulating soundfile ASTERISK-29714 Change-Id: I926ba4b11e9f6dd3fdd93170ab1f9b997910be70 |
3 years ago |
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a109b5aee0 |
pbx.c: Don't remove dashes from hints on reload.
When reloading dialplan, hints created dynamically would lose any dash characters. Now we ignore those dashes if we are dealing with a hint during a reload. ASTERISK-28040 #close Change-Id: I95e48f5a268efa3c6840ab69798525d3dce91636 |
4 years ago |
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5382b9dbb8 |
core: Don't play silence for Busy() and Congestion() applications.
When using the Busy() and Congestion() applications the function ast_safe_sleep is used by wait_for_hangup to safely wait on the channel. This function may send silence if Asterisk is configured to do so using the transmit_silence option. In a scenario where an answered channel dials a Local channel either directly or through call forwarding and the Busy() or Congestion() dialplan applications were executed with the transmit_silence option enabled the busy or congestion tone would not be heard. This is because inband generation of tones (such as busy and congestion) is stopped when other audio is sent to the channel they are being played to. In the given scenario the transmit_silence option would result in silence being sent to the channel, thus stopping the inband generation. This change adds a variant of ast_safe_sleep which can be used when silence should not be played to the channel. The wait_for_hangup function has been updated to use this resulting in the tones being generated as expected. ASTERISK-29485 Change-Id: I066bfc987a3ad6f0ccc88e0af4cd63f6a4729133 |
4 years ago |
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51cba591e3 |
pbx.c: On error, ast_add_extension2_lockopt should always free 'data'
In the event that the desired extension already exists, ast_add_extension2_lockopt() will free the 'data' it is passed before returning an error, so we should not be freeing it ourselves. Additionally, there were two places where ast_add_extension2_lockopt() could return an error without also freeing the 'data' pointer, so we add that. ASTERISK-29097 #close Change-Id: I904707aae55169feda050a5ed7c6793b53fe6eae |
5 years ago |
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28bae5e901 |
pbx: Fix hints deadlock between reload and ExtensionState.
When the ExtensionState AMI action is executed on a pattern matched hint it can end up adding a new hint if one does not already exist. This results in a locking order of contexts -> hints -> contexts. If at the same time a reload is occurring and adding its own hint it will have a locking order of hints -> contexts. This results in a deadlock as one thread wants a lock on contexts that the other has, and the other thread wants a lock on hints that the other has. This change enforces a hints -> contexts locking order by explicitly locking hints in the places where a hint is added when queried for. This matches the order seen through normal adding of hints. ASTERISK-29046 Change-Id: I49f027f4aab5d2d50855ae937bcf5e2fd8bfc504 |
5 years ago |
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3d1bf3c537 |
Compiler fixes for gcc 10
This patch fixes a few compile warnings/errors that now occur when using gcc 10+. Also, the Makefile.rules check to turn off partial inlining in gcc versions greater or equal to 8.2.1 had a bug where it only it only checked against versions with at least 3 numbers (ex: 8.2.1 vs 10). This patch now ensures any version above the specified version is correctly compared. Change-Id: I54718496eb0c3ce5bd6d427cd279a29e8d2825f9 |
5 years ago |
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50d02d6194 |
pbx.c: Include filesystem cache in free memory calculation
ASTERISK-28695 #close Reported by: Kevin Flyn Change-Id: Ief098bb6eb77378daeace8f97ba30701c8de55b8 |
5 years ago |
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bf6f27388d |
pbx: deadlock when outgoing dialed channel hangs up too quickly
Here's the basic scenario that occurred when executing an AMI fast originate while at the same time something else locks the channels container, and also wants a lock on the dialed channel: 1. pbx_outgoing_attempt obtains a lock on a dialed channel 2. concurrently another thread obtains a lock on the channels container, and subsequently requests a lock on the dialed channel. It waits on #1. For instance, "core show channel <dialed channel" 3. the outgoing call does not fail, but ends before the pbx_outgoing_attempt function exits 4. pbx_outgoing_attempt function exits, the outgoing structure destructs, and attempts to hang up the dialed channel 5. hang up tries to obtain the channels container lock, but can't due to #2. 6. Asterisk is deadlocked. The solution was to allow the pbx_outgoing_exec function to "steal" ownership of the dialed channel, and handle hanging it up. The channel now is either hung up prior to it being potentially locked by the initiating thread, or if locked the hang up takes place in a different thread, thus alleviating the deadlock. ASTERISK-28561 patches: iliketrains.diff submitted by Joshua Colp (license 5000) Change-Id: I51b42b92dde8f2215b69bb509e28667ee3a3853a |
6 years ago |
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702019fc80 |
pbx: Prevent Realtime switch crash on invalid priority
pbx_extension_helper takes two 'context' arguments. One (con) is a pointer directly to a 'struct ast_context' and the other (context) is the name of the context. In all cases, one of these arguments is NULL and the other is non-NULL. Functions that are ultimately called by pbx_extension_helper expect that 'context' will be non-NULL, so we set it unconditionally on entry into this function. ASTERISK-28534 #close Change-Id: Ifbbc5e71440afd80efd441f7a9d72e8b10b6f47d |
6 years ago |
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0afd6128a7 |
Merge "pbx.c: Ignore dashes in extensions when using extenpatternmatchnew"
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6 years ago |
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7e5709d726 |
pbx.c: Ignore dashes in extensions when using extenpatternmatchnew
Because hyphens are not matched literally in Asterisk dialplan, we need to ignore them in our candidate extensions as well. ASTERISK-17695 #close Reported by: test011 Change-Id: I227f02301577b1633e8a55b9fe9dc149935c03f0 |
6 years ago |
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2cf4e8bff9 |
pbx.c: Properly parse labels with leading digits
If the target of a Goto is a label that starts with a number, we erroneously treat the leading digits as a priority. ASTERISK-20182 #close Reported by: Janu Change-Id: Ia78408c0805a729103917247ecfc802f6fafc94b |
6 years ago |
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e2bbab17b3 |
Fix deadlock handling subscribe req during res_parking reload
Split destroy_hint method to separate hint removal and extension hint state changed callback, the latter now called via stasis. This avoids deadlock between res_parking reload that is removing the parking lot and the related hint and subscribe requests coming for the same parking lot. ASTERISK-28173 Change-Id: I5b03c3455b3b12b6f83cea4cc34f4b4b20444f7e |
6 years ago |
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021ce938ca
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astobj2: Remove legacy ao2_container_alloc routine.
Replace usage of ao2_container_alloc with ao2_container_alloc_hash or ao2_container_alloc_list. Remove ao2_container_alloc macro. Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088 |
7 years ago |
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3077ad0c24 |
stasis: Add internal filtering of messages.
This change adds the ability for subscriptions to indicate which message types they are interested in accepting. By doing so the filtering is done before being dispatched to the subscriber, reducing the amount of work that has to be done. This is optional and if a subscriber does not add message types they wish to accept and set the subscription to selective filtering the previous behavior is preserved and they receive all messages. There is also the ability to explicitly force the reception of all messages for cases such as AMI or ARI where a large number of messages are expected that are then generically converted into a different format. ASTERISK-28103 Change-Id: I99bee23895baa0a117985d51683f7963b77aa190 |
7 years ago |
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687ab7aeee
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astobj2: Eliminate legacy container allocation macros.
These macros have been documented as legacy for a long time but are still used in new code because they exist. Remove all references to: * ao2_container_alloc_options * ao2_t_container_alloc_options * ao2_t_container_alloc These macro's are also removed. Only ao2_container_alloc remains due to it's use in over 100 places. Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a |
7 years ago |
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b5914d90ac |
Fix GCC 8 build issues.
This fixes build warnings found by GCC 8. In some cases format truncation is intentional so the warning is just suppressed. ASTERISK-27824 #close Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84 |
7 years ago |
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c8a521b6c8 |
Replace direct checks of option_debug with DEBUG_ATLEAST macro.
Checking option_debug directly is incorrect as it ignores file/module specific debug settings. This system-wide change replaces nearly all direct checks for option_debug with the DEBUG_ATLEAST macro. Change-Id: Ic342d4799a945dbc40ac085ac142681094a4ebf0 |
7 years ago |
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77f2814d01 |
pbx: Reduce verbosity while loading extensions
Each time the dial plan is reloaded, a lot of logs like these are generated: "Added extension 'XXXXX' priority 1 to YYYYYYYYYYY" This patch changes the log level for those logs. ASTERISK-27084 Change-Id: I5662902161c50890997ddc56835d4cafb456c529 |
7 years ago |
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cfb88f3ac1 |
pbx: Prevent execution of NULL pointer.
pbx_extension_helper has a check for q->swo.exec == NULL but it doesn't actually return so we would still run the function. Fix the return. Move the 'int res' variable into the only scope which uses it. Also fix a copy-paste error in ast_pbx_init which could result in a crash on allocation failure (we exit with a normal error instead). Change-Id: I0693af921fdc7f56b6a72a21fb816ed08b960a69 |
7 years ago |
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fd0ca1c3f9 |
Remove as much trailing whitespace as possible.
Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0 |
7 years ago |
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5762f72425 |
PBX: Handle errors from AST_VECTOR_APPEND.
This resolves potentials leaks on AST_VECTOR_APPEND error in: * ast_context_add_include2 * ast_context_add_switch2 * ast_context_add_ignorepat2 Change-Id: Ib60e95c4f622fa3b832d87227c0523a695d736b6 |
8 years ago |
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325eeced6a |
core: Remove 'Data Retrieval API'
This API was not actively maintained, was not added to new modules (such as res_pjsip), and there exist better alternatives to acquire the same information, such as the ARI. Change-Id: I4b2185a83aeb74798b4ad43ff8f89f971096aa83 |
8 years ago |
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5a7af00e80 |
asterisk: Audit locking of channel when manipulating flags.
When manipulating flags on a channel the channel has to be locked to guarantee that nothing else is also manipulating the flags. This change introduces locking where necessary to guarantee this. It also adds helper functions that manipulate channel flags and lock to reduce repeated code. ASTERISK-26789 Change-Id: I489280662dba0f4c50981bfc5b5a7073fef2db10 |
8 years ago |
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c47b3e74d2 |
pbx: Use same thread if AST_OUTGOING_WAIT_COMPLETE specified
Both ast_pbx_outgoing_app() and ast_pbx_outgoing_exten() cause the core to spawn a new thread to perform the dial. When AST_OUTGOING_WAIT_COMPLETE is passed to these functions, the calling thread will be blocked until the newly created channel has been hung up. After this patch, we run the dial on the current thread rather than spawning a new one. The only in-tree code that passes AST_OUTGOING_WAIT_COMPLETE is pbx_spool, so you should see reduced thread usage if you are using .call files. Change-Id: I512735d243f0a9da2bcc128f7a96dece71f2d913 |
8 years ago |
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f997090877 |
pbx.c: Fix crash from malformed exten pattern.
Forgetting to indicate an exten is a pattern can cause a crash if the "pattern" has a character set range. e.g., "9999[3-5]" The crash is due to a buffer overwrite because the '-' exten eye-candy wasn't removed as expected and overran the allocated space. The buffer overwrite is fixed two ways in this patch. 1) Fix ext_strncpy() to distinguish between pattern and non-pattern extens. Now '-' characters are removed when they are eye-candy and not when they are part of a pattern character set. Since the function is private to pbx.c, the return value now returns the number of bytes written to the destination buffer instead of the strlen() of the final buffer so the callers that care don't need to add one. 2) Fix callers to ext_strncpy() to supply the correct available buffer size of the destination buffer. ASTERISK-26668 Change-Id: I555d97411140e47e0522684062d174fbe32aa84a |
8 years ago |
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c9296b23d1 |
core: Cleanup ast_get_hint() usage.
* manager.c:manager_state_cb() Fix potential use of uninitialized hint[] if a hint does not exist for the requested extension. Ran into this when developing a testsuite test. The AMI event ExtensionStatus came out with the hint header value containing garbage. The AMI event PresenceStatus also had the same issue. * manager.c:action_extensionstate() no need to completely initialize the hint[]. Only initialize the first element. * pbx.c:ast_add_hint() Remove unnecessary assignment. * chan_sip.c: Eliminate an unneeded hint[] local variable. We only care about the return value of ast_get_hint() there. Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b |
8 years ago |
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0910773077 |
manager: Restore Originate failure behavior from Asterisk 11
In Asterisk 11, if the 'Originate' AMI command failed to connect the provided
Channel while in extension mode, a 'failed' extension would be looked up and
run. This was, I believe, unintentionally removed in
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8 years ago |
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d96e350256 |
core/pbx: dialplan show - display filename/line#
Adds the ability for extensions to be registered to include filename and line number so that dialplan show output can show the filename and line number of a config file responsible for generating a given extension. This only affects config modules that are written to use the new extension registering functions. In this patch, that only includes pbx_config, so extensions registered in extensions.conf and any included extension will be shown in this manner. Extensions registered in this manner will show the filename and line number *instead* of the registrar. ASTERISK-26658 #close Reported by: Jonathan R. Rose Change-Id: Ieccc6abccdff34ed5c7da3511fd24972b8f2dd30 |
8 years ago |
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ddc951060a |
app_originate: Add option to execute gosub prior to dial
Issue/patch ASTERISK-26587 was inspired by issue ASTERISK-22992 that requested ability to add callerid into app_originate. Comments in that issue suggested that it was better solved by adding an option to gosub prior to originating the call. The attached patch implements this much like app_dial with two options one to gosub on the originating channel and one to gosub on the newly created channel and behaves just like app_dial. I have tested this patch by adding callerid info to the new channel and also SIPAddHeader (to e.g. add header to force auto answer) and confirmed it works. Have also tested both 'exten' and 'app' versions of app_originate. Opened by: dkerr Patch by: dkerr Change-Id: I36abc39b58567ffcab4a636ea196ef48be234c57 |
9 years ago |
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a6e5bae3ef |
Remove ASTERISK_REGISTER_FILE.
ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966 |
9 years ago |
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5e0758575c |
pbx.c: Prevent infinite recursion in manager_show_dialplan_helper.
Previously, if context A was including context B and context B was including context A, i.e. if there was a circular dependency between contexts, then calling manager_show_dialplan_helper could lead to an infinite recursion, resulting in a crash. This commit applies the same solution as the one implemented in the show_dialplan_helper function. The manager_show_dialplan_helper and show_dialplan_helper functions contain lots of code in common, but the former was missing the "infinite recursion avoidance" code. ASTERISK-26226 #close Change-Id: I1aea85133c21787226f4f8442253a93000aa0897 |
9 years ago |
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eca3d2698a |
pbx.c: Additional fixes to ast_context_remove_extension_callerid2.
Do not check registrar of the first extension head. We should only check the registrar when we match the priority. Additionally fix a couple calls to strcmp which used the input callerid instead of the clean version ex.cidmatch. ASTERISK-26233 Change-Id: I17ea6881a18f40840ae9c1f5394aab1fbb3769f1 |
9 years ago |
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b8fdd3ad79 |
Merge "pbx.c: Fix handling of '-' in extension name and callerid"
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9 years ago |