Before this patch, when a user hung up during a Background, we would
stuff 0xff into a char and attempt a dialplan lookup of it. This caused
problems for some realtime engines which interpreted the value as the
beginning of an invalid UTF-8 sequence.
ASTERISK-19291 #close
Reported by: Andrew Nowrot
Change-Id: I8ca6da93252d61c76ebdb46a4aa65e73ca985358
In review 4843 (ASTERISK-24858), we added a hack that forced a smoother
creation when sending signed linear so that the byte order was adjusted
during transmission. This was needed because smoother flags were lost
during the new format work that was done in Asterisk 13.
Rather than rolling that same hack into res_rtp_multicast, re-introduce
smoother flags so that formats can dictate their own options.
Change-Id: I77b835fba0e539c6ce50014a984766f63cab2c16
Rather than removing format_mp3 from ALL_C_MODS (which caused format_mp3
to not show up in menuselect), use .PHONY targets when the necessary
source files are not present.
ASTERISK-23951
Reported by: Tzafrir Cohen
Change-Id: I0a7512c51acc9e86043671795020b0de725bd9e8
Some of the test names were actually reserved words (true, false,
int, null, string, bool). When the jenkins test results analyzer
does its thing it tries to create a map using the test names as
keys and fails because they're reserved words.
Added "type_" to those test names.
Change-Id: I90d809f46969c78a1c605b736ff0635196a2cf1b
During the channel flag audit an incorrect change was
done. The flag should be cleared on the second channel.
ASTERISK-26469
Change-Id: I770c5a389550a2fb5a6ade942fccbb2e1d9199c8
...that can only be run by explicitly calling it with
'test execute category /DO_NOT_RUN/ name RAISE_SEGV'
This allows us to more easily test CI and debugging tools that
should do certain things when asterisk coredumps.
To allow this a new member was added to the ast_test_info
structure named 'explicit_only'. If set by a test, the test
will be skipped during a 'test execute all' or
'test execute category ...'.
Change-Id: Ia3a11856aae4887df9a02b6b081cc777b36eb6ed
This change allows the format of the EAGI audio pipe to be changed by
setting the dialplan variable 'EAGI_AUDIO_FORMAT' to the name of one of
the loaded formats.
ASTERISK-26124 #close
Change-Id: I7a10fad401ad2a21c68c2e7246fa357d5cee5bbd
Documented the 'beep' option in both the parameters list and the command
description.
ASTERISK-23839 #close
Change-Id: I4970395c922dbdce3f7cf0f56d5b065ec9aa53ea
Explicitly check that the appropriate number of arguments were passed to
SET VARIABLE before attempting to reference them. Also initialize the
arguments array to zeroes before populating it.
ASTERISK-22432 #close
Change-Id: I5143607d80a2724f749c1674f3126b04ed32ea97
If the generated XML documentation for a command does not end with a \n,
the postamble of the usage message does not appear on its own line.
ASTERISK-25662 #close
Change-Id: If190f1e9e37fe215fed95897d78d4a6e142b0020
Some devices separate format attributes with a semicolon followed by a
space, so trim blanks before trying to match them.
ASTERISK-27008 #close
Change-Id: Ia44cb2e4fef5c73dc541a29da79cb0e19c22d9cc
A change was done which added an 'in_call' flag to queue
members that was set to true while talking to an agent.
Unfortunately in practice this does not accurately reflect
whether they are talking to an agent or not. If a Local
channel is involved and a transfer is performed then the
app_queue application would incorrectly think the agent
was still in a call with the caller. This was done to
fix a race condition between an agent becoming available
by device state and the checking of the last call information
for the wrapup time. There was a small window where the
last call information would be the previous value instead
of the new one.
This change goes about fixing the original issue in a
different way by considering the call completed if device
state is received which would make the agent available
and if they are currently in a call. If this occurs the
last call information is updated before the agent becomes
available ensuring that old information is not present
when checking if the member should be called. This also
improves the transfer situation by actually updating
and enforcing the wrapup time.
ASTERISK-26399
ASTERISK-26400
ASTERISK-26715
ASTERISK-26975
Change-Id: Ife1cb686e3173b3a6d368601adef9aff69d4beea
When user leaves a conference, its channel calls async_play_sound_file()
in order to play the name announcement and then unlinks the sound file.
The async_play_sound_file() function adds a task to conference playback queue,
which then runs playback_common() function in a different thread.
It leads to a race condition when, in some cases, channel thread may unlink
the sound file before playback_common() had a chance to open it.
This patch creates a file deletion task, that is queued after playback.
ASTERISK-27012 #close
Change-Id: I412f7922d412004b80917d4e892546c15bd70dd3
When using rtcp mux if an rtcp payload came in it would still use the srtp
unprotect algorithm instead of the srtp unprotect rtcp method. Since rtcp
data was being passed to the rtp unprotect method this would result in an
error.
This patch ensures that the correct unprotect method is chosen by making
sure the passed in rtcp flag is appropriately set when rtcp mux is enabled
and an rtcp payload is received.
ASTERISK-26979 #close
Change-Id: Ic5409f9d1a267f1d4785fc5aed867daaecca6241
If we are offered or are offering RTCP-MUX, don't consider RTCP ICE
candidates. This confuses certain browsers (current Firefox for
example) and causes intial audio setup delays.
ASTERISK-26982 #close
Change-Id: Ifeaf47e83972fe8dbe58b7fb3d6d1823400cfb91
The while(1) loop in skinny_session wasn't checking for EOF so
a packet that was longer than a header but still truncated
would spin the while loop infinitely. Not only does this
permanently tie up a thread and drive a core to 100% utilization,
the call of ast_log() in such a tight loop eats all available
process memory.
Added poll with timeout to top of read loop
ASTERISK-26940 #close
Reported-by: Sandro Gauci
Change-Id: I2ce65f3c5cb24b4943a9f75b64d545a1e2cd2898
There are 2 places in app_queue.c that log EXITEMPTY event: one in
wait_our_turn, and another one in queue_exec in the loop trying to
call an agent after wait_our_turn.
In most cases it leads to logging EXITEMPTY twice.
ABANDON is also logged on two places, and in the rare case when an agent
and caller hang up simultaneously it's also possible to get duplicates
in queue_log.
This commit changes wait_our_turn to return -1 ("the caller should exit
the queue") instead of 0 ("the caller's turn has arrived") in case of
leaving when empty, so queue_exec skips the agent calling loop.
Also, leave_queue is now executed only once in this case, because 2nd
time is just a noop when the queue entry has already been removed.
Also, it sets qe->handled to -1 to indicate that the call was not
answered by an agent, but the necessary handling has already been done
in order to avoid logging an extra ABANDON entry.
ASTERISK-25665 #close
Reported by: Ove Aursand
Change-Id: I4578dd383bf2ac41589cf167865e8aaebcd4c11e
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
Retransmissions of an initial INVITE could be queued in the serializer
before we have processed the first INVITE message. If the first INVITE
message doesn't get completely processed before the retransmissions are
seen then we could try to setup the same call from the retransmissions. A
symptom of this is seeing a (key exists) message associated with an
INVITE. An earlier change attempted to address this kind of problem by
calculating a distributor serializer to use for unassociated messages.
Part of that change also made incoming calls keep using that distributor
serializer. (ASTERISK-26088) However, some leftover code was still
deferring the INVITE processing to the session's serializer even though we
were already in that serializer. This not only is unnecessary but would
cause the same call resetup problem.
* Removed the code to defer processing the initial INVITE to the session's
serializer because we are already running in that serializer.
ASTERISK-26998 #close
Change-Id: I1e822d82dcc650e508bc2d40d545d5de4f3421f6
