mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
5012 Commits (5d80a6e714fcf6dbd62a4720ea6853d8cb69a0b5)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
b2787876d6 |
main: Slight refactor of main. Improve color situation.
Several issues are addressed here: - main() is large, and half of it is only used if we're not rasterisk; fixed by spliting up the daemon part into a separate function. - Call ast_term_init from rasterisk as well. - Remove duplicate code reading/writing asterisk history file. - Attempt to tackle background color issues and color changes that occur. Tested by starting asterisk -c until the colors stopped changing at odd locations. ASTERISK-25585 #close Change-Id: Ib641a0964c59ef9fe6f59efa8ccb481a9580c52f |
10 years ago |
|
|
3f85a1be5a |
Merge "translate: Provide translation modules the result of SDP negotiation." into 13
|
10 years ago |
|
|
af288b2d96 |
main/cli: Use proper string methods to check existence of context/exten/app
Because the context, extension, and application are stored in stringfields, checking for them being NULL doesn't work so well. This patch uses the appropriate string library call, ast_strlen_zero, to see if there is a value in the context/exten/app values. Change-Id: Ie09623bfdf35f5a8d3b23dd596647fe3c97b9a23 |
10 years ago |
|
|
0b508789ab |
translate: Provide translation modules the result of SDP negotiation.
Previously, a trancoding module did not have access to the joint but cached format. Therefore, the module did not have access to the attributes negotiated via SDP (line fmtp). Now, a translation module receives the joint format. ASTERISK-25545 #close Change-Id: Id6878a989b50573298dab115d3371ea369e1a718 |
10 years ago |
|
|
e3cb27d341 |
Merge "format: Register format-attribute module with cached formats." into 13
|
10 years ago |
|
|
afd9a89e5a |
hashtab: Add NULL check when destroying iterator.
The hashtab API is pretty NULL tolerant which has resulted in remaining callers not doing much checks themselves. Unfortunately the function to destroy an iterator does not do a NULL check and will result in a crash if passed NULL. This change fixes that. ASTERISK-25552 #close Change-Id: Ic1bf8eec3639e5a440f1c941d3ae3893ac6ed619 |
10 years ago |
|
|
4f43b85c92 |
Taskprocessors: Increase high-water mark
In practical tests, we have seen certain taskprocessors, specifically Stasis subscription taskprocessors, cross the recently-added high-water mark and emit a warning. This high-water mark warning is only intended to be emitted when things have tanked on the system and things are heading south quickly. In the practical tests, the Stasis taskprocessors sometimes had a max depth of 180 tasks in them, and Asterisk wasn't in any danger at all. As such, this ups the high-water mark to 500 tasks instead. It also redefines the SIP threadpool request denial number to be a multiple of the taskprocessor high-water mark. Change-Id: Ic8d3e9497452fecd768ac427bb6f58aa616eebce |
10 years ago |
|
|
d8d3991390 |
format: Register format-attribute module with cached formats.
In Asterisk 13, cached formats are created before their corresponding format- attribute module is registered. Cached formats are involved when a local extension is called. Therefore, ast_format_generate_sdp_fmtp did not work on local extensions. This change affects the Opus Codec, H.263 (Plus), H.264, and format-attribute modules provided externally. ASTERISK-25160 #close Change-Id: I1ea1f0483e5261e2a050112e4ebdfc22057d1354 |
10 years ago |
|
|
cd51b0aeac |
Merge "res_pjsip: Deny requests when threadpool queue is backed up." into 13
|
10 years ago |
|
|
2f9cb7d62b |
res_pjsip: Deny requests when threadpool queue is backed up.
We have observed situations where the SIP threadpool may become deadlocked. However, because incoming traffic is still arriving, the SIP threadpool's queue can continue to grow, eventually running the system out of memory. This change makes it so that incoming traffic gets rejected with a 503 response if the queue is backed up too much. Change-Id: I4e736d48a2ba79fd1f8056c0dcd330e38e6a3816 |
10 years ago |
|
|
4e5bf12b33 |
format_cap: Don't append the 'none' format when appending all.
When appending all formats of a type all the codecs are iterated and added. This operation was incorrectly adding the ast_format_none format which is special in that it is supposed to be used when no format is present. It shouldn't be appended. ASTERISK-25535 Change-Id: I7b00f3bdf4a5f3022e483d6ece602b1e8b12827c |
10 years ago |
|
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b818d70533 |
threadpool: Handle worker thread transitioning to dead when going active.
This change adds handling of dead worker threads when moving them to be active. When this happens the worker thread is removed from both the active and idle threads container. If no threads are able to be moved to active then the pool grows as configured. A unit test has also been added which thrashes the idle timeout and thread activation to exploit any race conditions between the two. ASTERISK-25546 #close Change-Id: I6c455f9a40de60d9e86458d447b548fb52ba1143 |
10 years ago |
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dac0bf063c |
Merge "rtp_engine: Init a format-attribute module to its RFC defaults." into 13
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10 years ago |
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e098fb1813 |
Merge "ast_format_cap: Avoid format creation on module load, use cache instead." into 13
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10 years ago |
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bd157b9ca8 |
Merge "xmldoc: Improve xmldoc wrapping of 'core show ...' output." into 13
|
10 years ago |
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4bf84459c7 |
rtp_engine: Init a format-attribute module to its RFC defaults.
Previously, format-attribute modules relied on an existing fmtp line in SDP negotiation. However, fmtp is optional for several formats like the Opus Codec. Now, the format-attribute module is called with an empty fmtp, which allows the module to initialise itself to RFC defaults. Furthermore now, Asterisk is able to differentiate between internally and externally created formats. ASTERISK-25537 #close Change-Id: I28f680cef7fdf51c0969ff8da71548edad72ec52 |
10 years ago |
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18e61a6442 |
Merge "taskprocessor: Add high water mark warnings" into 13
|
10 years ago |
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1bff400df7 |
ast_format_cap_get_names: To display all formats, the buffer was increased.
ASTERISK-25533 #close Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a |
10 years ago |
|
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f3ac4d8090 |
ast_format_cap: Avoid format creation on module load, use cache instead.
Since Asterisk 13, formats are immutable and cached. However while loading a module like chan_sip, some formats were created instead using cached ones. ASTERISK-25535 #close Change-Id: I479cdc220d5617c840a98f3389b3bd91e91fbd9b |
10 years ago |
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0d425f2eb4 |
xmldoc: Improve xmldoc wrapping of 'core show ...' output.
Previously, the wrapping did both lookahead and lookback, which, together with color escape sequences, caused some lines to be wrapped way earlier than other lines. This led to inconsistent output. This simplifies the wrapping code and makes it more sane: if maxcolumns is hit, we simply jump back to the last space and wrap there. ASTERISK-25527 #close Change-Id: I56d01c6f9a812642b1b05535c98d4db48d17c957 |
10 years ago |
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6ff48319d9 |
taskprocessor: Add high water mark warnings
If a taskprocessor's queue grows large, this can indicate that there may be a problem with tasks not leaving the processor or else that the number of available task processors for a given type of task is too low. This patch makes it so that if a taskprocessor's task queue grows above 100 queued tasks that it will emit a warning message. Warning messages are emitted only once per task processor. ASTERISK-25518 #close Reported by: Jonathan Rose Change-Id: Ib1607c35d18c1d6a0575b3f0e3ff5d932fd6600c |
10 years ago |
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506aea26e6 |
main/dial: Protect access to the format_cap structure of the requesting channel
When a dial attempt is made that involves a requesting channel, we previously were not: a) Protecting access to the native format capabilities structure on the requesting channel. That is inherently unsafe. b) Reference bumping the lifetime of the format capabilities structure. In both cases, something else could sneak in, blow away the format capabilities, and we'd be holding onto an invalid format_cap structure. When the newly created channel attempts to construct its format capabilities, things go poorly. This patch: a) Ensures that we get a reference to the native format capabilities while the requesting channel is locked b) Holds a reference to the native format capabilities during the creation of the new channel. ASTERISK-25522 #close Change-Id: I0bfb7ba8b9711f4158cbeaae96edf9626e88a54f |
10 years ago |
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d098d00424 |
Fix cli display of build options.
A previous commit reduced the AST_BUILDOPTS compiler define to only include options that affected ABI. This included some options that were previously displayed by cli "core show settings". This change corrects the CLI display while still restricting buildopts.h to ABI effecting options only. ASTERISK-25434 #close Reported by: Rusty Newton Change-Id: Id07af6bedd1d7d325878023e403fbd9d3607e325 |
10 years ago |
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f0f190af08 |
main/stasis_endpoints: Fix ContactStatusChange JSON for roundtrip_usec field
The JSON packing for the ContactStatusChange event forgot to include the roundtrip_usec field. As a result, the field never showed up in any event, even when the data was available. This patch corrects that error by properly packing the JSON blob with the data. Change-Id: I8df80da659a44010afbd48f645967518ff5daa17 |
10 years ago |
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7d1a2b0839 |
Merge "format: Update the maximum packetization time for iLBC 30." into 13
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10 years ago |
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84f1068bab |
Merge topic 'fix_oom_crash' into 13
* changes: strings.c: Fix __ast_str_helper() to always return a terminated string. Add missing failure checks to ast_str_set_va() callers. |
10 years ago |
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ebe69dee0d |
format_cap: Detect vector allocation failures.
A crash was seen on a system that ran out of memory due to Asterisk not checking for vector allocation failures in format_cap.c. With this change, if either of the AST_VECTOR_INIT calls fail, we will return a value indicating failure. Change-Id: Ieb9c59f39dfde6d11797a92b45e0cf8ac5722bc8 |
10 years ago |
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c8c65dfa41 |
strings.c: Fix __ast_str_helper() to always return a terminated string.
Users of functions which call __ast_str_helper() such as the ones listed below are likely to not check the return value for failure so ensuring that the string is always nil terminated is a good safety measure. ast_str_set_va() ast_str_append_va() ast_str_set() ast_str_append() Change-Id: I36ab2d14bb6015868b49329dda8639d70fbcae07 |
10 years ago |
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b271d4a28a |
Add missing failure checks to ast_str_set_va() callers.
Change-Id: I0c2cdcd53727bdc6634095c61294807255bd278f |
10 years ago |
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84ff075d41 |
format: Update the maximum packetization time for iLBC 30.
In September 2006, the maximum packetization time (ptime) were set to such a low value, packetization was disabled for many codecs actually. This was fixed for many codecs but not for iLBC 30. This enables packetization for iLBC which can be enabled for example via allow=ilbc:60,gsm,alaw,ulaw in the file sip.conf. ASTERISK-7803 Change-Id: I2ef90023d35efb7cb8fe96ed74f53f6846ffad12 |
10 years ago |
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e14023ca35 |
config.c: Fix off-nominal memory leak.
Change-Id: I06e346e9a5c63cc5071e7eda537310c4b43bffe0 |
11 years ago |
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a99e821520 |
config.c: Fix potential memory corruption after [section](+).
The memory corruption could happen if the [section](+) is the last section in the file with trailing comments. In this case process_text_line() has left *last_cat is set to newcat and newcat is destroyed. Change-Id: I0d1d999f553986f591becd000e7cc6ddfb978d93 |
11 years ago |
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8d31d2526b |
config.c: Fix #include after [section](+).
An #include right after a [section](+) would associate any variable assignments before a new section in the #include with the wrong section. * Fix section association by setting the current section to the appended section. * Fix '+' and '!' section flag interaction corner case depending upon which flag came first. If the '!' came first then it would be ignored. If the '!' came after then it would affect the appended section. The '!' will now no longer be ignored. ASTERISK-25461 #close Reported by: Sean Pimental Change-Id: Ic9d3191c8758048e2cbce6432f854b32531731c3 |
11 years ago |
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74635b5638 |
manager: Fix GetConfigJSON returning invalid JSON
When GetConfigJSON was introduced back in 1.6, it returned each
section as an array of strings: ["key=value", "key2=value2"].
Afterwards, it was changed a few times and became
["key": "value", "key2": "value2"], which is not a correct JSON.
This patch fixes that by constructing a JSON object {} instead of
an array [].
Also, the keys "istemplate" and "tempates" that are used to
indicate templates and their inherited categories are now wrapped in
quotes.
ASTERISK-25391 #close
Reported by: Bojan Nemčić
Change-Id: Ibbe93c6a227dff14d4a54b0d152341857bcf6ad8
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11 years ago |
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ddebb217f0 |
sched.c: Add warning about negative time interval request.
Change-Id: Ib91435fb45b7f5f7c0fc83d0eec20b88098707bc |
11 years ago |
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771757e338 |
Merge "translate: Fix transcoding while different in frame size." into 13
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11 years ago |
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41f856e5a2 |
Merge "logger: Prevent duplicate dynamic channels from being added." into 13
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11 years ago |
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f050fa76eb |
logger: Prevent duplicate dynamic channels from being added.
There was a problem observed where the "logger add channel" CLI command would allow for a channel with the same name to be added multiple times. This would result in each message being written out to the same file multiple times. The problem was due to the difference in how logger channel filenames are stored versus the format they are allowed to be presented when they are added. For instance, if adding the logger channel "foo" through the CLI, the result would be a logger channel with the file name /var/log/asterisk/foo being stored. So when trying to add another "foo" channel, "foo" would not match "/var/log/asterisk/foo" so we'd happily add the duplicate channel. The fix presented here is to introduce two new methods in the logger code: * make_filename(): given a logger channel name, this creates the filename for that logger channel. * find_logchannel(): given a logger channel name, this calls make_filename() and then traverses the list of logchannels in order to find a match. This change has made use of make_filename() and find_logchannel() throughout to more consistently behave. ASTERISK-25305 #close Reported by Mark Michelson Change-Id: I892d52954d6007d8bc453c3cbdd9235dec9c4a36 |
11 years ago |
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629458d349 |
Do not swallow frames on channels leaving bridges.
When leaving a bridge, indications on a channel could be swallowed by the internal indication logic because it appears that the channel is on its way to be hung up anyway. One such situation where this is detrimental is when channels on hold are redirected out of a bridge. The AST_CONTROL_UNHOLD indication from the bridging code is swallowed, leaving the channel in question to still appear to be on hold. The fix here is to modify the logic inside ast_indicate_data() to not drop the indication if the channel is simply leaving a bridge. This way, channels on hold redirected out of a bridge revert to their expected "in use" state after the redirection. ASTERISK-25418 #close Reported by Mark Michelson Change-Id: If6115204dfa0551c050974ee138fabd15f978949 |
11 years ago |
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0d95610ae2 |
Merge "ARI: Add events for Contact and Peer Status changes" into 13
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11 years ago |
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5f15cd93f0 |
app_page.c: Fix crash when forwarding with a predial handler.
Page uses the async method of dialing with the dial API. When a call gets forwarded there is no calling channel available. If the predial handler was set then the calling channel could not be put into auto-service for the forwarded call because it doesn't exist. A crash is the result. * Moved the callee predial parameter string processing to before the string is passed to the dial API rather than having the dial API do it. There are a few benefits do doing this. The first is the predial parameter string processing doesn't need to be done for each channel called by the dial API. The second is in async mode and the forwarded channel is to have the predial handler executed on it then the non-existent calling channel does not need to be present to process the predial parameter string. * Don't start auto-service on a non-existent calling channel to execute the predial handler when the dial API is in async mode and forwarding a call. ASTERISK-25384 #close Reported by: Chet Stevens Change-Id: If53892b286d29f6cf955e2545b03dcffa2610981 |
11 years ago |
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b50e372394 |
ARI: Add events for Contact and Peer Status changes
This patch adds support for receiving events regarding Peer status changes and Contact status changes. This is particularly useful in scenarios where we are subscribed to all endpoints and channels, where we often want to know more about the state of channel technology specific items than a single endpoint's state. ASTERISK-24870 Change-Id: I6137459cdc25ce27efc134ad58abf065653da4e9 |
11 years ago |
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ec514ad64d |
core/logging: Fix logging to more than one syslog channel
Currently, Asterisk will log to the last configured syslog channel in logger.conf. This is due to the fact that the final call to openlog() supersedes all of the previous calls. This commit removes the call to openlog() and passes the facility to ast_log_vsyslog(), along with utilizing the LOG_MAKEPRI macro to ensure that the message is routed to the correct facility and with the correct priority. ASTERISK-25407 #close Reported by: Elazar Broad Tested by: Elazar Broad Change-Id: Ie2a2416bc00cce1b04e99ef40917c2011953ddd2 |
11 years ago |
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a45abbf3ce |
Merge "pbx: Update device and presence state when changing a hint extension." into 13
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11 years ago |
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960f00939a |
Merge "main/config_options: Check for existance of internal object before derefing" into 13
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11 years ago |
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2bd27d1222 |
pbx: Update device and presence state when changing a hint extension.
When changing a hint extension without removing the hint first the device state and presence state is not updated. This causes the state of the hint to be that of the previous extension and not the current one. This state is kept until a state change occurs as a result of something (presence state change, device state change). This change updates the hint with the current device and presence state of the new extension when it is changed. Any state callbacks which may have been added before the hint extension is changed are also informed of the new device and presence state if either have changed. ASTERISK-25394 #close Change-Id: If268f1110290e502c73dd289c9e7e7b27bc8432f |
11 years ago |
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b88c54fa4b |
translate: Fix transcoding while different in frame size.
When Asterisk translates between codecs, each with a different frame size (for example between iLBC 30 and Speex-WB), too large frames were created by ast_trans_frameout. Now, ast_trans_frameout is called with the correct frame length, creating several frames when necessary. Affects all transcoding modules which used ast_trans_frameout: GSM, iLBC, LPC10, and Speex. ASTERISK-25353 #close Change-Id: I2e229569d73191d66a4e43fef35432db24000212 |
11 years ago |
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5c713fdf18 |
scheduler: Use queue for allocating sched IDs.
It has been observed that on long-running busy systems, a scheduler context can eventually hit INT_MAX for its assigned IDs and end up overflowing into a very low negative number. When this occurs, this can result in odd behaviors, because a negative return is interpreted by callers as being a failure. However, the item actually was successfully scheduled. The result may be that a freed item remains in the scheduler, resulting in a crash at some point in the future. The scheduler can overflow because every time that an item is added to the scheduler, a counter is bumped and that counter's current value is assigned as the new item's ID. This patch introduces a new method for assigning scheduler IDs. Instead of assigning from a counter, a queue of available IDs is maintained. When assigning a new ID, an ID is pulled from the queue. When a scheduler item is released, its ID is pushed back onto the queue. This way, IDs may be reused when they become available, and the growth of ID numbers is directly related to concurrent activity within a scheduler context rather than the uptime of the system. Change-Id: I532708eef8f669d823457d7fefdad9a6078b99b2 |
11 years ago |
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4eedd9ef9d |
main/config_options: Check for existance of internal object before derefing
Asterisk can load and register an object type while still having an invalid sorcery mapping. This can cause an issue when a creation call is invoked. For example, mis-configuring PJSIP's endpoint identifier by IP address mapping in sorcery.conf will cause the sorcery mechanism to be invalidated; however, a subsequent ARI invocation to create the object will cause a crash, as the internal type may not be registered as sorcery expects. Merely checking for a NULL pointer here solves the issue. Change-Id: I54079fb94a1440992f4735a9a1bbf1abb1c601ac |
11 years ago |
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4ed9c9a280 |
Core/General: Add #ifdef needed on FreeBSD.
pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED on FreeBSD too. ASTERISK-25310 #close Reported by: Guido Falsi Change-Id: Iae6befac9028b5b9795f86986a4a08a1ae6ab7c4 |
11 years ago |