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Change-Id: Ic1f9e22ba1f2ff3b3f5cb017c5ddcd9bd48eccc7
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ASTERISK-20987 #close
Reported by: hristo
Change-Id: Ic61a2b524ab3a4cfadf227fc6b3506527bc03f38
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Since res_pjsip now depends on res_pjproject, this has been added to
basic-pbx modules.conf.
Change-Id: I42826597d5e10f08e518208860c44c96e52f1b2d
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A recent change to queue channel variable setting to the Stasis control
queue caused a regression. When setting channel variables, it is
possible to give a NULL channel variable value in order to unset the
variable (i.e. remove it from the channel variable list). The change
introduced a call to ast_variable_new(), which is not tolerant of NULL
channel variable values.
This new change switches from using ast_variable to using a custom
channel variable struct that is lighter weight and NULL value-tolerant.
Change-Id: I784d7beaaa3c036ea936d103e7caf0bb1562162d
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Core and extra sounds 1.5 was recently released! The tarballs contain
change descriptions however I figure more people will see this one so
I'll try to be a bit detailed. Approximately 60 sounds were moved from Extra
to Core for en, en_GB, fr and added for languages that didn't already
have Extra sound sets (it,ja,ru).
In addition all of the English and Russian sounds have been completely
re-recorded.
Sounds moved and added:
activated,added,all-circuits-busy-now,astcc-followed-by-pound
at-tone-time-exactly,call-forwarding,call-fwd-no-ans,call-fwd-on-busy
,call-fwd-unconditional,calling,call-waiting,cancelled,
cannot-complete-as-dialed,check-number-dial-again,conf-full,de-activated
,disabled,do-not-disturb,enabled,enter-num-blacklist,entr-num-rmv-blklist
,extension,feature-not-avail-line,for,from-unknown-caller,goodbye,hello
,if-correct-press,im-sorry,info-about-last-call,is,is-in-use,is-set-to
,location,number,number-not-answering,num-was-successfully,one-moment-please
,please-try-again,pls-hold-while-try,pls-try-call-later,pm-invalid-option
,privacy-to-blacklist-last-caller,removed,simul-call-limit-reached
,something-terribly-wrong,sorry,sorry-youre-having-problems,speed-dial
,speed-dial-empty,telephone-number,time,to-call-this-number,to-extension
,to-listen-to-it,to-rerecord-it,unidentified-no-callback,with,you-entered
,your
There were also a few random fixes here and there to file names for a few
of the languages.
ASTERISK-25068 #close
Change-Id: I2b594344ec585d7dfd922b40c1af43b1508828b3
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A test recently uncovered that running an ill-timed AMI command to show
inbound subscriptions could cause a crash since Asterisk will try to
operate on a freed subscription.
The fix for this is to remove the subscription tree from the list of
subscriptions at the time that we are sending our final NOTIFY request
out. This way, as the subscription is in the process of dying, it is
inaccessible from AMI.
Change-Id: Ic0239003d8d73e04c47c12dd2a7e23867e5b5b23
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sip_sipredirect uses sscanf to copy up to 256 characters to a stacked buffer
of 256 characters. This patch reduces the copy to 255 characters to leave
room for the string null terminator.
ASTERISK-25722 #close
Change-Id: Id6c3a629a609e94153287512c59aa1923e8a03ab
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When queuing tasks onto the Stasis control queue, you can pass an
arbitrary data pointer and a function to free that data. All ARI
commands that use the Stasis control queue made the assumption that the
destructor function would be called in all paths, whether the task was
queued successfully or not. However, this was not correct. If a task was
queued onto a control structure that was already completed, the
allocated data would not be freed properly.
This patch corrects this by making sure that all return paths call the
data destructor.
Change-Id: Ibf06522094f8e5c4cce652537dc5d7222b1c4fcb
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A crash occurred when attempting to set a channel variable on a channel
that had already been hung up. This is because there is a small window
between when a control is grabbed and when the channel variable is set
that the channel can be hung up.
The fix here is to queue the setting of the channel variable onto the
control queue. This way, the manipulation of the channel happens in a
thread where it is safe to be done.
In this change, I also noticed that the setting of bridge roles on
channels was being done outside of the control queue, so I also changed
those operations to be done in the control queue.
ASTERISK-25709 #close
Reported by Mark Michelson
Change-Id: I2a0a4d51bce6fba6f1d9954e40935e42f366ea78
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The null terminator of the tail struct member was not being allocated
when no logger.conf config file is installed.
ASTERISK-25714 #close
Reported by: Badalian Vyacheslav
Change-Id: I45770fdd08af39506a3bc33ba279c4f16e047a30
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* Add 'check-alembic' target to root Makefile.
* Create build_tools/make_check_alembic to do the actual checks.
ASTERISK-25685
Change-Id: Ibb3cae7d1202ac23dc70b0f3b5801571ad46b004
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ASTERISK-25712 #close
Reported by: Richard Mudgett
Change-Id: I70634df24f8c6c3a2c66c45af61d021e4999253f
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Dump the res_pjsip endpt internals.
In non-developer mode we will not document or make easily accessible the
"details" option even though it is still available. The user has to know
it exists to use it. Presumably they would also be aware of the potential
crash warning below.
Warning: PJPROJECT documents that the function used by this CLI command
may cause a crash when asking for details because it tries to access all
active memory pools.
Change-Id: If2d98a3641c9873364d1daaad971376311aef3cb
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Make sure buf[res] is not accessed at res=-1 (buffer underrun).
Address Sanitizer will complain about this quite loudly.
ASTERISK-24801 #close
Change-Id: Ifcd7f691310815a31756b76067c56fba299d3ae9
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res_pjsip_log_forwarder has been renamed to res_pjproject
and enhanced as follows:
As a follow-on to the recent 'Add CLI "pjsip show buildopts"' patch,
a new ast_pjproject_get_buildopt function has been added. It
allows the caller to get the value of one of the buildopts.
The initial use case is retrieving the runtime value of
PJ_MAX_HOSTNAME to insure we don't send a hostname greater
than pjproject can handle. Since it can differ between
the version of pjproject that Asterisk was compiled against
and the version of pjproject that Asterisk is running against,
we can't use the PJ_MAX_HOSTNAME macro directly in Asterisk
source code.
Change-Id: Iab6e82fec3d7cf00c1cf6185c42be3e7569dee1e
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and billsec"
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When CDRs were refactored, func_cdr's ability to report high precision values
for duration and billsec (the 'f' option) was broken. This was due to func_cdr
incorrectly interpreting the duration/billsec values provided by the CDR engine
in milliseconds, as opposed to seconds. Since the CDR engine only provides
duration and billsec in seconds, and does not expose either attribute with
sufficient precision to merely pass back the underlying value, this patch fixes
the bug by re-calculating duration and billsec with microsecond precision based
on the start/answer/end times on the CDR.
ASTERISK-25179 #close
Change-Id: I8bc63822b496537a5bf80baf6102c06206bee841
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This change makes the thread_timeout_thrash unit test wait for
each task to complete. This fixes the problem where the test would
prematurely end when all threads were gone and a new one had to be
started to handle the last task. It also increases the thrasing as
it is now more likely for each task to encounter the above scenario.
This also fixes a memory leak where the data for each task was not
being freed.
ASTERISK-25611 #close
Change-Id: I5017d621a4dc911f509074c16229b86bff2fb3c6
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Change-Id: I4892d6acbb580d6c207d006341eaf5e0f8f2a029
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You have to call ast_taskprocessor_unref() outside of the taskprocessor
implementation code. Taskprocessor use since v12 has become more
transient than just the singleton uses in earlier versions.
Change-Id: If7675299924c0cc65f2a43a85254e6f06f2d61bb
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Change-Id: I64ed21fec50eb833641ca49d92184f6aaabd86e8
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This removes logchannels locking from init_logger_chain, puts the
responsibility on the caller. Adds locking around the one call that was
missing it.
ASTERISK-24833
Change-Id: I6cc42117338bf9575650a67bcb78ab1a33d7bad8
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Change-Id: I5387821f29e5caa0cba0b7d62b0fc0d341e7e20b
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Update the version number in the comments from Asterisk 12 to Asterisk 12+
Change-Id: Ie692ac8cda3c993c3bf10f27f51a1cca3317ec7b
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Corrects the qualify_timeout column type from Integer to Decimal
ASTERISK-25686 #close
Reported-by: Marcelo Terres
Change-Id: I757d0e3c011ee9be6cd5abd48bc92441a405d3c8
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hangs up"
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ASTERISK-25700 #close
Change-Id: I096da84f9c62c6095f68bcf98eac4b7c7868e808
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The xferfailsound was read from the channel at the beginning of the transfer,
and that value is "cached" for the duration of the transfer. Therefore, changing
the xferfailsound on the channel using the FEATURE() dialplan function does
nothing once the transfer is under way.
This makes it so the transfer code instead gets the xferfailsound configuration
options from the channel when it is actually going to be used.
This patch also fixes a potential memory leak of the props object as well as
making sure the condition variable gets initialized before being destroyed.
ASTERISK-25696 #close
Change-Id: Ic726b0f54ef588bd9c9c67f4b0e4d787934f85e4
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Change-Id: Id5bd18ef1f60ef8be453e677e98478298358a9d1
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* Add freed regions totals to allocations and summary.
* Add totals for all allocations and not just the selected allocations.
Change-Id: I61d5a5112617b0733097f2545a3006a344b4032a
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If the attended transfer destination answers (picks call up or goes to
voicemail) and then hangs up on the transferer then transferer hears the
fail sound.
This patch makes it so the fail sound is not played when the transfer
destination/target hangs up after answering.
ASTERISK-25697 #close
Change-Id: I97f142fe4fc2805d1a24b7c16143069dc03d9ded
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