We were ignoring the return value from ast_pbx_outgoing_exten() and
ast_pbx_outgoing_app() which could fail before setting the reason code.
This resulted in failures being reported as success.
ASTERISK-25266 #close
Reported by: Allen Ford
Change-Id: Idf16237b7e41b527d2c69c865829128686beeb3b
The previous patch for ASTERISK-27216 made it so you wouldn't get any
position or periodic announcements unless you had announce-to-first-user
enabled. The announce-to-first-user feature was added by ASTERISK_21782
as a result of the patch which introduced the redundant announcements that
ASTERISK-27216 removes.
* By noting that the makeannouncement variable is used to suppresses the
first user announcement, we set its initial value to the
announce-to-first-user enable setting.
ASTERISK-27216
Change-Id: Ieaeb7dbea8ae7073086b775fbafe0625b000b10a
This patch reverts the change by patch 2263 from old reviewboard.
Note that reverting that 2263-patch still preserves the behaviour that
the commit log of the 2263-patch claimed to add. The reason for this is:
The function wait_for_answer is only called from try_calling which
in turn is only called from the main for loop in queue_exec, and
earlier in that loop we already check the things that's removed by
this patch. There's no need to check those things twice each loop
iteration, and I think the proper place to check it is before each
ringing cycle. By checking it in wait_for_answer, you allow the issue
explained in the jira - that the head caller hears announcements while
the agents' sip phones are actively ringing.
Reported-by: Stefan Engström
Tested-by: Stefan Engström
ASTERISK-27216 #close
Change-Id: Ic4290dc75256f9743900c6762ee1bb915f672db0
* WaitForSilence completes successfully if it receives no media in the
specified timeout, but when acting as WaitForNoise that logic needs
to be reversed.
* Use standard argument parsing macros and add some error checking for
invalid values.
* The documentation indicated that the first argument to both
WaitForSilence and WaitForNoise was required when it was not. Update
the documentation to reflect that.
* Wrap up some behavior in structs to avoid boolean checks all over the
place.
ASTERISK-24066 #close
Reported by: M vd S
Change-Id: I01d40adc5b63342bb5018a1bea2081a0aa191ef9
An admin can configure app_minivm with an externnotify program to be run
when a voicemail is received. The app_minivm application MinivmNotify
uses ast_safe_system() for this purpose which is vulnerable to command
injection since the Caller-ID name and number values given to externnotify
can come from an external untrusted source.
* Add ast_safe_execvp() function. This gives modules the ability to run
external commands with greater safety compared to ast_safe_system().
Specifically when some parameters are filled by untrusted sources the new
function does not allow malicious input to break argument encoding. This
may be of particular concern where CALLERID(name) or CALLERID(num) may be
used as a parameter to a script run by ast_safe_system() which could
potentially allow arbitrary command execution.
* Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp()
instead of ast_safe_system() to avoid command injection.
* Document code injection potential from untrusted data sources for other
shell commands that are under user control.
ASTERISK-27103
Change-Id: I7552472247a84cde24e1358aaf64af160107aef1
This prevents orphaned CBAnn channels from getting stuck in the bridge.
ASTERISK-26994 #close
Reported by: James Terhune
Change-Id: I5e43e832a9507ec3f2c59752cd900b41dab80457
mkstemp() returns a unique filename, but appending an extension to that
filename does not guarantee uniqueness. Instead, use mkdtemp() and we
can put whatever extension we want on the files that we create inside
the directory.
In the case of app_minivm, we also now properly clean up any temporary
files that we create.
ASTERISK-20858 #close
Reported by: Walter Doekes
Change-Id: I30ad04f0e115f0b11693ff678ba5184d8b938e43
If the Record() application is called with a relative filename that
includes directories, we were not properly creating the intermediate
directories and Record() would fail.
Secondarily, updated the documentation for RECORDED_FILE to mention
that it does not include a filename extension.
Finally, rewrote the '%d' functionality to be a bit more straight
forward and less noisy.
ASTERISK-16777 #close
Reported by: klaus3000
Change-Id: Ibc2640cba3a8c7f17d97b02f76b7608b1e7ffde2
Fixed to use correct initial value and fixed to use the
correct queue info to check the first value.
ASTERISK-27204
Change-Id: Ia9e36c828e566e1cc25c66f73307566e4acb8e73
GCC 7 has added capability to produce warnings, this fixes most of those
warnings. The specific warnings are disabled in a few places:
* app_voicemail.c: truncation of paths more than 4096 chars in many places.
* chan_mgcp.c: callid truncated to 80 chars.
* cdr.c: two userfields are combined to cdr copy, fix would break ABI.
* tcptls.c: ignore use of deprecated method SSLv3_client_method().
ASTERISK-27156 #close
Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88
The following testsuite voicemail tests were failing to re-enter the
mailbox after the first login attempt.
tests/apps/voicemail/authenticate_invalid_mailbox
tests/apps/voicemail/authenticate_invalid_password
The tests were noting the start of the vm-incorrect-mailbox prompt and
immediately sending the mailbox for the next login attempt. Since the
invalid message playback had to complete before the digits were
recognized, the test passed for the wrong reason and added approximately
20 seconds to the test times.
* Allow the vm-incorrect-mailbox prompt to get interrupted by the mailbox
digits like the initial vm-login prompt so the tests are able to enter the
intended mailbox.
Change-Id: I1dc53fe917bfe03a4587b2c4cd24c94696a69df8
This commit fixes two possible scenarios:
* When recording name and if during recording you hangup, file is never
removed. This is due to the fact file location is nulled.
* When recording name and if you hangup during thank-you prompt, file
is never removed.
ASTERISK-27123 #close
Change-Id: I39b7271408b4b54ce880c5111a886aa8f28c2625
In say_date_generic the timezonename parameter is passed but never
used. Fix it by passing it to the ast_localtime function.
ASTERISK-27124
Change-Id: I6afa98f9163190043244b9f3ba91eb1874d1b586
The primary focus of this patch is adding a missing call to
ast_odbc_release_obj(), but is also a general cleanup of the ODBC
related code in app_voicemail.
ASTERISK-27093 #close
Change-Id: I8e285142eaeb3146b4287a928276b70db76c902b
The fix for ASTERISK-25665 introduced a regression.
The return value of queue_exec used to be 0 in case of leavewhenempty
but it was changed to -1 (returned from wait_our_turn and passed
transparently by queue_exec), thus leading to hangup instead of returning
back to dialplan.
This commit resets the value back to 0 in this case, restoring
original behavior.
ASTERISK-27065 #close
Reported by: Marek Cervenka
Change-Id: Id9c83b75aeda463250155e88c5004be52bbca5ac
A new global option "imap_poll_logout" was added to specify whether need to
disconnect from the IMAP server after polling of mailboxes.
ASTERISK-27068 #close
Closing IMAP connection after loading mailbox from voicemail.conf
ASTERISK-24052 #close
Change-Id: Ib7558ba04516240a32b65f42e9be64372a0ae12a
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
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
menu_template_handler wasn't properly accounting for the fact that
it might be called both during a load/reload (which isn't really
valid but not prevented) and by a dialplan function. In both cases
it was attempting to use the "pending" config which wasn't valid in
the latter case. aco_process_config is also partly to blame because
it wasn't properly cleaning "pending" up when a reload was done and
no changes were made. Both of these contributed to a crash if
CONFBRIDGE(menu,template) was called in a dialplan after a reload.
* aco_process_config now sets info->internal->pending to NULL
after it unrefs it although this isn't strictly necessary in the
context of this fix.
* menu_template_handler now uses the "current" config and silently
ignores any attempt to be called as a result of someone uses the
"template" parameter in the conf file.
Luckily there's no other place in the codebase where
aco_pending_config is used outside of aco_process_config.
ASTERISK-25506 #close
Reported-by: Frederic LE FOLL
Change-Id: Ib349a17d3d088f092480b19addd7122fcaac21a7
Cleaned up some of the incorrect uses of fread() and fwrite(), mostly in
the format modules. Neither of these functions will ever return a value
less than 0, which we were checking for in some cases.
I've introduced a fair amount of duplication in the format modules, but
I plan to change how format modules work internally in a subsequent
patch set, so this is simply a stop-gap.
Change-Id: I8ca1cd47c20b2c0b72088bd13b9046f6977aa872
In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed
to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting
if a module can't be loaded. If the user wishes to retain the
FAILURE behavior for a specific module, they can use the "require"
or "preload-require" keyword in modules.conf.
A new API was added to logger: ast_is_logger_initialized(). This
allows asterisk.c/check_init() to print to the error log once the
logger subsystem is ready instead of just to stdout. If something
does fail before the logger is initialized, we now print to stderr
instead of stdout.
Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25
Queue member will get stuck in pending_members if queue calls a device
that is different from the one observed for state changes.
This patch removes members from pending_members as a result of channel stasis
events such as blind or attended transfers and hangup.
ASTERISK-26862 #close
Change-Id: I8bf6df487b9bb35726c08049ff25cdad5e357727
The queue_stasis_data structure contains various mutable fields that require
appropriate locking. Specifically, the 'dying,' 'member_uniqueid,' and
'caller_uniqueid' fields need to be locked when read from or written to.
Change-Id: I246b7dbff8447acc957a1299f6ad0ebd0fd39088
