mirror of https://github.com/asterisk/asterisk
master
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${ noResults }
5084 Commits (c1abe8906dd4696bc85697cfbbff2c9fca6af4d6)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
c1abe8906d |
Merge "rtp_engine: Init a format-attribute module to its RFC defaults."
|
10 years ago |
|
|
9c9ebbd482 |
Merge "dns: Use ntohl for ans->ttl in dns_parse_answer_ex"
|
10 years ago |
|
|
7987bf54c6 |
Merge "ast_format_cap: Avoid format creation on module load, use cache instead."
|
10 years ago |
|
|
8b4b3e9b88 |
Merge "xmldoc: Improve xmldoc wrapping of 'core show ...' output."
|
10 years ago |
|
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525c7ab780 |
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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d8dadf4894 |
Merge "taskprocessor: Add high water mark warnings"
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10 years ago |
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e4af14784e |
Merge "Remove ABI compatibility stub functions."
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10 years ago |
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0be147f713 |
Merge "ast_format_cap_get_names: To display all formats, the buffer was increased."
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10 years ago |
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be93036a4e |
Remove ABI compatibility stub functions.
ABI compatibility stubs existed for ast_app_separate_args and ast_verbose, this is not needed in master. Change-Id: I07b4d2c16079da3c2c6efa55df4a74368e0bd453 |
10 years ago |
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02a124eda5 |
Remove execute permission from dns_system_resolver.c
Change-Id: I3185735db42064bab00d3e073aed703385a00bf4 |
10 years ago |
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cf79b62778 |
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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e85f0c81af |
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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d82a4b098f |
dns: Use ntohl for ans->ttl in dns_parse_answer_ex
dns_parse_answer_ex was not converting ans->ttl from network by order to host byte order which was causing certain ttls it to go negative. In turn this was causing answer edit checks to fail. ASTERISK-25528 #close Reported-by: Daniel Tryba Tested-by: George Joseph Change-Id: I31505132d6321c46d2f39fd06c20ee808a864037 |
10 years ago |
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74e7333317 |
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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a2c2a8e1bb |
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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9c293b5104 |
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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b0bf189908 |
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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e26a06c1da |
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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6b1e9fbdcf |
Merge "format: Update the maximum packetization time for iLBC 30."
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10 years ago |
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d73bd56b0a |
Merge topic 'fix_oom_crash'
* 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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5dd9e1938a |
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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1ce62b2545 |
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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a04d946eaa |
Add missing failure checks to ast_str_set_va() callers.
Change-Id: I0c2cdcd53727bdc6634095c61294807255bd278f |
10 years ago |
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f9cbac7321 |
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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77780790e0 |
main/cdr: Allow modules to modify CDR fields before dispatching them
This patch adds the functions ast_cdr_modifier_register() ast_cdr_modifier_unregister() That work much like ast_cdr_register() and ast_cdr_unregister(). Modules registered will be given a chance to modify (or to do whatever they want) CDR fields just before they are passed to registered engines. Thus, for instance, if a module change the "userfield" field of a CDR, the modified value will be passed to every registered CDR backend for logging. ASTERISK-25479 #close Change-Id: If11d8fd19ef89b1a66ecacf1201e10fcf86ccd56 |
10 years ago |
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984f100dab |
config.c: Fix off-nominal memory leak.
Change-Id: I06e346e9a5c63cc5071e7eda537310c4b43bffe0 |
10 years ago |
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9951255775 |
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 |
10 years ago |
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c1ed11ee31 |
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 |
10 years ago |
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89dec7675d |
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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10 years ago |
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9bc7386b7c |
sched.c: Add warning about negative time interval request.
Change-Id: Ib91435fb45b7f5f7c0fc83d0eec20b88098707bc |
11 years ago |
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2d7a4a3357 |
main/logger: Add log formatters and JSON structured logs
When Asterisk is part of a larger distributed system, log files are often
gathered using tools (such as logstash) that prefer to consume information
and have it rendered using other tools (such as Kibana) that prefer a
structured format, e.g., JSON. This patch adds support for JSON formatted
logs by adding support for an optional log format specifier in Asterisk's
logging subsystem. By adding a format specifier of '[json]':
full => [json]debug,verbose,notice,warning,error
Log messages will be output to the 'full' channel in the following
format:
{
"hostname": Hostname or name specified in asterisk.conf
"timestamp": Date/Time
"identifiers": {
"lwp": Thread ID,
"callid": Call Identifier
}
"logmsg": {
"location": {
"filename": Name of the file that generated the log statement
"function": Function that generated the log statement
"line": Line number that called the logging function
}
"level": Log level, e.g., DEBUG, VERBOSE, etc.
"message": Actual text of the log message
}
}
ASTERISK-25425 #close
Change-Id: I8649bfedf3fb7bf3138008cc11565553209cc238
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11 years ago |
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df7cfc9ac9 |
Merge "translate: Fix transcoding while different in frame size."
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11 years ago |
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6392fdf6dc |
Merge "logger: Prevent duplicate dynamic channels from being added."
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11 years ago |
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3eefa07a39 |
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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f42084be09 |
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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6f719bb4d0 |
Merge "ARI: Add events for Contact and Peer Status changes"
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11 years ago |
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06f4f80a63 |
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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069813db3c |
Merge "core/logging: Fix logging to more than one syslog channel"
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11 years ago |
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4effba0d0a |
Merge "pbx: Update device and presence state when changing a hint extension."
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11 years ago |
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4c2b77618c |
Merge "astfd: Adds a timestamp for each entry."
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11 years ago |
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a29cf45c76 |
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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5206aa9d30 |
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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5541c6de6e |
Merge "main/config_options: Check for existance of internal object before derefing"
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11 years ago |
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9200ad03a3 |
astfd: Adds a timestamp for each entry.
Now with menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", a timestamp is shown with each file descriptor. This helps to debug leaked UDP/TCP ports on long-lived servers, for example. ASTERISK-25405 #close Change-Id: I968339e5155a512eba1032a5263f1ec8b5e1f80b |
11 years ago |
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42a897c4c3 |
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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077adf48b8 |
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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0a74c80300 |
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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45cf79665c |
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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fdf77633ed |
Merge "Core/General: Add #ifdef needed on FreeBSD."
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11 years ago |
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7691035312 |
endpoint snapshot: avoid second cleanup on alloc failure
In ast_endpoint_snapshot_create(), a failure to init the string fields results in two attempts to ao2_cleanup the same pointer. Removed RAII_VAR to eliminate problem. ASTERISK-25375 #close Reported by: Scott Griepentrog Change-Id: If4d9dfb1bbe3836b623642ec690b6d49b25e8979 |
11 years ago |