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PJSIP name formats:
pjsip/aor/<aor>-<seq> -- registrar thread pool serializer
pjsip/default-<seq> -- default thread pool serializer
pjsip/messaging -- messaging thread pool serializer
pjsip/outreg/<registration>-<seq> -- outbound registration thread pool
serializer
pjsip/pubsub/<endpoint>-<seq> -- pubsub thread pool serializer
pjsip/refer/<endpoint>-<seq> -- REFER thread pool serializer
pjsip/session/<endpoint>-<seq> -- session thread pool serializer
pjsip/websocket-<seq> -- websocket thread pool serializer
Change-Id: Iff9df8da3ddae1132cb2ef65f64df0c465c5e084
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Sorcery name formats:
sorcery/<type>-<seq> -- Sorcery thread pool serializer
Change-Id: Idc2e5d3dbab15c825b97c38c028319a0d2315c47
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Stasis name formats:
subm:<topic>-<seq> -- Stasis subscription mailbox task processor
subp:<topic>-<seq> -- Stasis subscription thread pool serializer
Change-Id: Id19234b306e3594530bb040bc95d977f18ac7bfd
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* 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
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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
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Change-Id: I1d9f4e532caa6dfabe034745dd16d06134efdce5
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Change-Id: I71e7bf57c7b908c8b8c71f1816348ed7c5a5d51e
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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
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Change-Id: I78247e0faf978bf850b5ba4e9f4933ab3c59d17b
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Fix compile error in main/utils.c because strdup was used in dummy_start
Change-Id: Id61a6cf4f3cbf235450441e10e7da101a6335793
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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
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into 13
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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
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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
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* 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.
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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
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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
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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
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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
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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
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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
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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
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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
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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
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into 13
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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
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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
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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
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last one" into 13
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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
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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
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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
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into 13
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debugging" into 13
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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
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write errors" into 13
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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
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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
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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
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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 b9bd3c14, certain network
conditions could cause the header to be written, and then the
sub-sequent body to fail - which would cause the next successful
write to contain a new header, and a new body (resulting in
the peer receiving two headers - the second of which would be
read as part of the body for the first header).
This was patched to have both write operations individually fail
by closing the websocket.
In a case available to the submitter of this patch, the same
body which would consistently fail to write, would succeed
if written at the same time as the header.
This update merges the two operations in to one, adds debug messages
indicating the reason for a websocket connection being closed during
a write operation, and clarifies some variable names for code legibility.
Change-Id: I4db7a586af1c7a57184c31d3d55bf146f1a40598
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deleted" into 13
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Change-Id: I9d88fac0394d5bbaff0900a2ee911c4e4478846b
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