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1.4.26-rc5
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1.4.27
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1.4.27-rc2
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1.4.42
1.4.42-rc1
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1.4.7
1.4.7.1
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1.4.9
1.6.0
1.6.0-beta1
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1.6.0.10
1.6.0.11-rc1
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1.6.0.12
1.6.0.13
1.6.0.13-rc1
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1.6.0.14-rc1
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1.6.0.16
1.6.0.16-rc1
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1.6.0.17
1.6.0.18
1.6.0.18-rc1
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1.6.0.19
1.6.0.2
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1.6.0.21
1.6.0.21-rc1
1.6.0.22
1.6.0.23
1.6.0.23-rc1
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1.6.0.28
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1.6.0.3
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1.6.0.3.1
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1.6.0.5
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10.12.0
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10.12.2
10.12.2-digiumphones
10.12.3
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10.12.4
10.12.4-digiumphones
10.2.0
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10.2.1
10.3.0
10.3.0-rc1
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10.3.1
10.4.0
10.4.0-digiumphones-rc1
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10.4.0-rc1
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10.5.0
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10.5.0-digiumphones-rc1
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10.5.0-rc1
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10.5.1
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10.5.2
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10.6.0
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10.7.0
10.7.0-digiumphones
10.7.0-digiumphones-rc1
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10.8.0
10.8.0-digiumphones
10.8.0-digiumphones-rc1
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10.8.0-rc1
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10.9.0
10.9.0-digiumphones
10.9.0-digiumphones-rc1
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11.0.0
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11.1.1
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11.2.1
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11.20.0-rc1
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11.21.0
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11.5.1
11.6-cert11
11.6.0
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11.6.1
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12.0.0
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12.5.1
12.6.0
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12.7.1
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13.0.0
13.0.0-beta1
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13.0.1
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13.1-cert2
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13.1.1
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certified/1.8.11-cert1
certified/1.8.11-cert10
certified/1.8.11-cert2
certified/1.8.11-cert3-rc1
certified/1.8.11-cert3-rc2
certified/1.8.11-cert4
certified/1.8.11-cert5
certified/1.8.11-cert5-rc1
certified/1.8.11-cert5-rc2
certified/1.8.11-cert6
certified/1.8.11-cert7
certified/1.8.11-cert8
certified/1.8.11-cert9
certified/1.8.11-cert9-rc1
certified/1.8.15-cert1
certified/1.8.15-cert1-rc1
certified/1.8.15-cert1-rc2
certified/1.8.15-cert1-rc3
certified/1.8.15-cert2
certified/1.8.15-cert3
certified/1.8.15-cert4
certified/1.8.15-cert5
certified/1.8.15-cert6
certified/1.8.15-cert7
certified/1.8.28-cert1
certified/1.8.28-cert1-rc1
certified/1.8.28-cert2
certified/1.8.28-cert3
certified/1.8.28-cert4
certified/1.8.28-cert5
certified/1.8.6-cert1
certified/11.2-cert1
certified/11.2-cert1-rc1
certified/11.2-cert1-rc2
certified/11.2-cert2
certified/11.2-cert3
certified/11.6-cert1
certified/11.6-cert1-rc1
certified/11.6-cert1-rc2
certified/11.6-cert10
certified/11.6-cert11
certified/11.6-cert12
certified/11.6-cert2
certified/11.6-cert3
certified/11.6-cert4
certified/11.6-cert5
certified/11.6-cert6
certified/11.6-cert7
certified/11.6-cert8
certified/11.6-cert9
certified/13.1-cert1
certified/13.1-cert1-rc1
certified/13.1-cert1-rc2
certified/13.1-cert1-rc3
certified/13.1-cert2
certified/13.1-cert3
${ noResults }
27284 Commits (ea9deff996744b41ae383876b6c4e64936d85a9c)
Author | SHA1 | Message | Date |
---|---|---|---|
|
4276f185f0 |
Sorcery: Create human friendly serializer names.
Sorcery name formats: sorcery/<type>-<seq> -- Sorcery thread pool serializer Change-Id: Idc2e5d3dbab15c825b97c38c028319a0d2315c47 |
10 years ago |
|
f02ac1b7f9 |
Stasis: Create human friendly taskprocessor/serializer names.
Stasis name formats: subm:<topic>-<seq> -- Stasis subscription mailbox task processor subp:<topic>-<seq> -- Stasis subscription thread pool serializer Change-Id: Id19234b306e3594530bb040bc95d977f18ac7bfd |
10 years ago |
|
ec1f1c6742 |
taskprocessor.c: New API for human friendly taskprocessor names.
* Add new API call to get a sequence number for use in human friendly taskprocessor names. * Add new API call to create a taskprocessor name in a given buffer and append a sequence number. Change-Id: Iac458f05b45232315ed64aa31b1df05b875537a9 |
10 years ago |
|
d8bc3e0c8b |
taskprocessor.c: Fix CLI "core show taskprocessors" output format.
Update the CLI "core show taskprocessors" output format to not be distorted because UUID names are longer than previously used taskprocessor names. Change-Id: I1a5c82ce3e8f765a0627796aba87f8f7be077601 |
10 years ago |
|
2c4b7502de |
taskprocessor.c: Fix CLI "core show taskprocessors" unref.
Change-Id: I1d9f4e532caa6dfabe034745dd16d06134efdce5 |
10 years ago |
|
3b33ac7a46 |
taskprocessor.c: Sort CLI "core show taskprocessors" output.
Change-Id: I71e7bf57c7b908c8b8c71f1816348ed7c5a5d51e |
10 years ago |
|
0fc32c4dd3 |
ccss.c: Replace space in taskprocessor name.
The CLI "core ping taskprocessor" command does not work very well with taskprocessor names that have spaces in them. You have to put quotes around the name so using tab completion becomes awkward. Change-Id: I29e806dd0a8a0256f4e2e0a7ab88c9e19ab0eda0 |
10 years ago |
|
0e0c24ad78 |
taskprocessor.c: Add CLI "core ping taskprocessor" missing unlock.
Change-Id: I78247e0faf978bf850b5ba4e9f4933ab3c59d17b |
10 years ago |
|
0f79c8839b |
main: Use ast_strdup instead of strdup
Fix compile error in main/utils.c because strdup was used in dummy_start Change-Id: Id61a6cf4f3cbf235450441e10e7da101a6335793 |
10 years ago |
|
4285dee778 |
include/asterisk/time.h: Renamed global declaration:tv
Renamed global declaration:tv to dummy_tv_var_for_types, which would oltherwise cause 'shadow' warnings when 'tv' was declared as a local variable elsewhere. Added comment to note that dummy_tv_var_for_types is never really exported and only used as a place holder. ASTERISK-25627 #close Change-Id: I9a6e17995006584f3627efe8988e3f8aa0f5dc28 |
10 years ago |
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01e5894388 |
Merge "PJSIP: Prevent deadlock due to dialog/transaction lock inversion." into 13
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10 years ago |
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96094feab6 |
PJSIP: Prevent deadlock due to dialog/transaction lock inversion.
A deadlock was observed where the monitor thread was stuck, therefore resulting in no incoming SIP traffic being processed. The problem occurred when two 200 OK responses arrived in response to a terminating NOTIFY request sent from Asterisk. The first 200 OK was dispatched to a threadpool worker, who locked the corresponding transaction. The second 200 OK arrived, resulting in the monitor thread locking the dialog. At this point, the two threads are at odds, because the monitor thread attempts to lock the transaction, and the threadpool thread loops attempting to try to lock the dialog. In this case, the fix is to not have the monitor thread attempt to hold both the dialog and transaction locks at the same time. Instead, we release the dialog lock before attempting to lock the transaction. There have also been some debug messages added to the process in an attempt to make it more clear what is going on in the process. ASTERISK-25668 #close Reported by Mark Michelson Change-Id: I4db0705f1403737b4360e33a8e6276805d086d4a |
10 years ago |
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52e9de0016 |
ast_format_cap_append_by_type: Resolve codec reference leak.
This resolves a reference leak caused by ASTERISK-25535. The pointer returned by ast_format_get_codec is saved so it can be released. ASTERISK-25664 #close Change-Id: If9941b1bf4320b2c59056546d6bce9422726d1ec |
10 years ago |
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eb3d083551 |
Merge "cel/cel_radius: Fix wrong pointer." into 13
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10 years ago |
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b1ee692568 |
Merge topic 'pbx-split' into 13
* changes: main/pbx: Move hangup handler routines to pbx_hangup_handler.c. main/pbx: Move dialplan application management routines to pbx_app.c. main/pbx: Move switch routines to pbx_switch.c. |
10 years ago |
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2e61c3d97b |
Merge "main/pbx: Move timing routines to pbx_timing.c." into 13
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10 years ago |
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86eae38d7e |
cel/cel_radius: Fix wrong pointer.
The macro ADD_VENDOR_CODE defined in the cel_radius.c should use the parameter y not the address of y. I capture the radius UDP packet via tcpdump, and the AV pairs are not correct, then i review the source code and compare it with cdr/cdr_radius.c. Fix it and it works. ASTERISK-25647 #close Reported by: Aaron An Tested by: Aaron An Change-Id: I72889bccd8fde120d47aa659edc0e7e6d4d019f0 |
10 years ago |
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881dc862e0 |
asterisk.h: Add ASTERISK_REGISTER_FILE macro
The 11/13 branches and master use 2 different file version macros. 11/13 uses ASTERISK_FILE_VERSION but master uses ASTERISK_REGISTER_FILE. This means a new file added to 11/13 can't just be cherry-picked to master because the macro has to be changed. To make cherry-picking possible, ASTERISK_REGISTER_FILE was added to asterisk.h as a simple alias for ASTERISK_FILE_VERSION(__FILE__, NULL) The "$Revision$" tag doesn't do anything since Asterisk moved to git so just passing NULL as the verison works fine. asterisk.h was also annotated to deprecate ASTERISK_FILE_VERSION and suggest using ASTERISK_REGISTER_FILE for all new files. Finally, 2 recent file additions, pbx_builtins.c and pbx_functions.c, were modified to use the new macro to make sure it actually worked. 'core show file version' showed the correct output. Change-Id: I5867ed898818d26ee49bb6e5c7d4c1a45d4789a5 |
10 years ago |
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9016e51c03 |
Merge "main/pbx: Move variable routines to pbx_variables.c." into 13
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10 years ago |
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d228b62fd4 |
stasis_cache_pattern: Backport to 13
Somehow stasis_cache_pattern got out of sync between 13 and master and it was causing duplicate channel message issues in 13 when related to a specific endpoint. I.E. from statsd, 'endpoints.PJSIP.1174.channels 0|g' was being emitted twice. Backporting stasis_cache_pattern from master to 13 solved the issue and running the unit and testsuite tests confirmed that no new ones were created. ASTERISK-25317 #close Change-Id: Ia8707462f62d15eed14541c37f332a7bbbceb548 |
10 years ago |
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e462f0063f |
main/pbx: Move hangup handler routines to pbx_hangup_handler.c.
This is the sixth patch in a series meant to reduce the bulk of pbx.c. This moves hangup handler management functions to their own source. Change-Id: Ib25a75aa57fc7d5c4294479e5cc46775912fb104 |
10 years ago |
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ab191d124c |
main/pbx: Move dialplan application management routines to pbx_app.c.
This is the sixth patch in a series meant to reduce the bulk of pbx.c. This moves dialplan application management functions to their own source. Change-Id: I444c10fb90a3cdf9f3047605d6a8aad49c22c44c |
10 years ago |
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09a9b93896 |
main/pbx: Move switch routines to pbx_switch.c.
This is the fifth patch in a series meant to reduce the bulk of pbx.c. This moves ast_switch functions to their own source. Change-Id: Ic2592a18a5c4d8a3c2dcf9786c9a6f650a8c628e |
10 years ago |
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c608274a39 |
main/pbx: Move timing routines to pbx_timing.c.
This is the fourth patch in a series meant to reduce the bulk of pbx.c. This moves pbx timing functions to their own source. Change-Id: I05c45186cb11edfc901e95f6be4e6a8abf129cd6 |
10 years ago |
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338a8ffed6 |
app_queue: Add member flag "in_call" to prevent reading wrong lastcall time
Member lastcall time is updated later than member status. There was chance to check wrapuptime for available member with wrong (old) lastcall time. New boolean flag "in_call" is set to true right before connecting call, and reset to false after update of lastcall time. Members with "in_call" set to true are treat as unavailable. ASTERISK-19820 #close Change-Id: I1923230cf9859ee51563a8ed420a0628b4d2e500 |
10 years ago |
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e13719bff1 |
app_queue: Added reason pause of member
In app_queue added value Paused Reason on QueueMemberStatus when a member on queue is paused and the reason was set. ASTERISK-25480 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: Ia5db503482f50764c15e2020196c785f59d4a68e |
10 years ago |
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937ba51659 |
Merge "voicemail: Move app_voicemail / res_mwi_external conflict to runtime" into 13
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10 years ago |
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4ec85a9f07 |
voicemail: Move app_voicemail / res_mwi_external conflict to runtime
The menuselect conflict between app_voicemail and res_mwi_external makes it hard to package 1 version of Asterisk. There no actual build dependencies between the 2 so moving this check to runtime seems like a better solution. The ast_vm_register and ast_vm_greeter_register functions in app.c were modified to return AST_MODULE_LOAD_DECLINE instead of -1 if there is already a voicemail module registered. The modules' load_module functions were then modified to return DECLINE instead of -1 to the loader. Since -1 is interpreted by the loader as AST_MODULE_LOAD_FAILURE, the modules were incorrectly causing Asterisk to stop so this needed to be cleaned up anyway. Now you can build both and use modules.conf to decide which voicemail implementation to load. The default menuselect options still build app_voicemail and not res_mwi_external but if both ARE built, res_mwi_external will load first and become the voicemail provider unless modules.conf rules prevent it. This is noted in CHANGES. Change-Id: I7d98d4e8a3b87b8df9e51c2608f0da6ddfb89247 |
10 years ago |
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7fdcfd7724 |
main/pbx: Move variable routines to pbx_variables.c.
This is the third patch in a series meant to reduce the bulk of pbx.c. This moves channel and global variable routines to their own source. Change-Id: Ibe8fb4647db11598591d443a99e3f99200a56bc6 |
10 years ago |
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80a8b2a4cd |
app_dial: Immediately exit dial if the caller is already hung up.
If a caller hangs up before dial is executed within an AGI then the AGI has likely eaten all queued frames before executing the dial in DeadAGI mode. With the caller hung up and no pending frames from the caller's read queue, dial would not know that the call has hung up until a called channel answers. It is rather annoying to whoever just answered the non-existent call. Dial should not continue execution in DeadAGI mode, hangup handlers, or the h exten. * Added a check early in dial to abort dialing if the caller has hungup. ASTERISK-25307 #close Reported by: David Cunningham Change-Id: Icd1bc0764726ef8c809f76743ca008d0f102f418 |
10 years ago |
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f42036bf6b |
Merge "main/cdr: Allow setting properties on a finalized CDR if it is the last one" into 13
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10 years ago |
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4e1d9c4a20 |
Merge "main/cdr: Set the end time on a CDR if endbeforehexten is Yes" into 13
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10 years ago |
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1087b0c6ed |
main/cdr: Allow setting properties on a finalized CDR if it is the last one
Prior to this patch, we explicitly disallowed setting any properties on a finalized CDR. This seemed like a good idea at the time; in practice, it was more restrictive. There are weird and strange scenarios where setting a property on a finalized CDR is definitely wrong. For example, we may Fork a CDR, finalizing the previous one, then change a property. In said case, the old CDR is supposed to now be 'immutable' (so to speak), and should not be updated. From the perspective of the code, a forked CDR that is finalized is just finalized. Hence why we decided these should not be updated. In practice, it is much more common to want to set a property on a CDR in the h extension or in a hangup handler. Disallowing a common scenario to make an esoteric behaviour work isn't good. This patch fixes this by allowing callers to set a property IF we are the last CDR in the chain. This preserves the finalized CDR if it was forked, while allowing the more common case to function. ASTERISK-25458 #close Change-Id: Icf3553c607b9f561152a41e6d8381d594ccdf4b9 |
10 years ago |
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1f23e65b89 |
main/cdr: Set the end time on a CDR if endbeforehexten is Yes
Prior to this patch, the CDR engine attempted to set the end time on a CDR that was executing hangup logic and with endbeforehexten set to Yes by calling a function that inspects the properties on the Party A snapshot to determine if we are ready to set the end time. That always failed. This is because a Party A snapshot is not updated for CDRs that are executing hangup logic with endbeforehexten=Yes. Instead of calling a function that looks at the Party A snapshot, we just simply set the end time on the CDR. This is safe to call multiple times, and is safe to call at this point as we know that (a) we are executing hangup logic, and (b) we are supposed to set the end time at this point. ASTERISK-25458 Change-Id: I0c27b493861f9c13c43addbbb21257f79047a3b3 |
10 years ago |
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2ffade4574 |
main/pbx: Move custom function routines to pbx_functions.c.
This is the second patch in a series meant to reduce the bulk of pbx.c. This moves custom function management routines to their own source. Change-Id: I34a6190282f781cdbbd3ce9d3adeac3c3805e177 |
10 years ago |
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bc7c882326 |
Merge "main/pbx: Move pbx_builtin dialplan applications to pbx_builtins.c" into 13
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10 years ago |
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5a75caa9e6 |
Merge "res_pjsip_history: Add a module that provides PJSIP history for debugging" into 13
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10 years ago |
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20b8474f20 |
main/pbx: Move pbx_builtin dialplan applications to pbx_builtins.c
We joked about splitting pbx.c into multiple files but this first step was fairly easy. All of the pbx_builtin dialplan applications have been moved into pbx_builtins.c and a new pbx_private.h file was added. load_pbx_builtins() is called by asterisk.c just after load_pbx(). A few functions were renamed and are cross-exposed between the 2 source files. Change-Id: I87066be3dbf7f5822942ac1449d98cc43fc7561a |
10 years ago |
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a68467d293 |
Merge "res_http_websocket.c: prevent avoidable disconnections caused by write errors" into 13
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10 years ago |
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47d508e911 |
Merge "test_time: Provide a timeout when waiting." into 13
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10 years ago |
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e4a566918a |
tests/test_stasis_endpoints: Remove expected duplicate events
The cache_clear test was written to expect duplicate Stasis messages sent from the technology endpoint to the all caching topic. This patch fixes the test to no longer expect these duplicate messages. ASTERISK-25137 Change-Id: I58075d70d6cdf42e792e0fb63ba624720bfce981 |
10 years ago |
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a280400758 |
test_time: Provide a timeout when waiting.
The test_timezone_watch unit test is written to expect a condition to be signaled when the inotify daemon thread runs. There exists a small window where the test_timezone_watch thread can signal the inotify daemon thread while it is not reading on the underlying file descriptor. If this occurs the test_timezone_watch thread will wait indefinitely for a signal that will never arrive. This change adds a timeout to the condition so it will return regardless after a period of time. Change-Id: Ifed981879df6de3d93acd3ee0a70f92546517390 |
10 years ago |
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3a1c4885be |
endpoint/stasis: Eliminate duplicate events on endpoint status change
When an endpoint is created, its messages are forwarded to both the tech endpoint topic and the all endpoints topic. This is done so that various parties interested in endpoint messages can subscribe to just the tech endpoint and receive all messages associated with that particular technology, as opposed to subscribing to the all endpoints topic. Unfortunately, when the tech endpoint is created, it also forwards all of its messages to the all topic. This results in duplicate messages whenever an endpoint publishes its messages. This patch resolves the duplicate message issue by creating a new function for Stasis caching topics, stasis_cp_sink_create. In most respects, this acts as a normal caching topic, save that it no longer forwards messages it receives to the all endpoints topic. This allows it to act as an aggregation "sink", while preserving the necessary caching behaviour. ASTERISK-25137 #close Reported-by: Vitezslav Novy ASTERISK-25116 #close Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com> Change-Id: Ie47784adfb973ab0063e59fc18f390d7dd26d17b |
10 years ago |
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136c537695 |
res_http_websocket.c: prevent avoidable disconnections caused by write errors
Updated ast_websocket_write to encode the entire frame in to one
write operation, to ensure that we don't end up with a situation
where the websocket header has been sent, while the body can not
be written.
Previous to August's patch in commit
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10 years ago |
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5e99ba393e |
Merge "bridge_basic.c: Fix GOTO_ON_BLINDXFR" into 13
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10 years ago |
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1a1f64f36f |
Merge "chan_sip.c: fix websocket_write_timeout default value" into 13
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10 years ago |
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efa0868295 |
Merge "res/res_pjsip_location: Delete contact_status object when contact is deleted" into 13
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10 years ago |
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84b1475674 |
Merge "Remove res_jabber file that was left behind." into 13
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10 years ago |
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f2efbb5d75 |
Remove res_jabber file that was left behind.
Change-Id: I9d88fac0394d5bbaff0900a2ee911c4e4478846b |
10 years ago |
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dde7f3c1c4 |
res_pjsip_history: Add a module that provides PJSIP history for debugging
This patch adds a new module, res_pjsip_history, that provides a slightly better way of debugging SIP message traffic on a busy Asterisk system. The existing mechanisms all rely on passively dumping a SIP message to the CLI. While this is perfectly fine for logging purposes and well controlled environments, on many installations, the amount of SIP messages Asterisk receives will quickly swamp the CLI. This makes it difficult to view/capture those messages that you want to diagnose in real time. This patch provides another way of handling this. When enabled, the module will store SIP message traffic in memory. This traffic can then be queried at leisure. In order to make the querying useful, a CLI command has been implemented, 'pjsip show history', that supports a basic expression syntax similar to SQL or other query languages. A small number of useful fields have been added in this initial patch; additional fields can easily be added in later improvements. Those fields are: - number: The entry index in the history - timestamp: The time the message was recieved - addr: The source/destination address of the message - sip.msg.request.method: The request method - sip.msg.call-id: The Call-ID header Note - this is a resurrection of the module initially proposed on Review Board here: https://reviewboard.asterisk.org/r/4053/ Change-Id: I39bd74ce998e99ad5ebc0aab3e84df3a150f8e36 |
10 years ago |