Age | Commit message (Collapse) | Author |
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If a db_sync was scheduled just before shutdown, the atexit code calling
db_sync would have no effect, causing the astdb commit thread to stay
alive. This caused the SIP/realtime_sipregs test to fail. (The fallback
kill would run the atexit code again and that would wreak havoc.) This
fixes that the atexit kill condition is picked up properly.
(closes issue ASTERISK-18883)
Reviewed by: Terry Wilson
Review: https://reviewboard.asterisk.org/r/1659
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The Asterisk -x command line parameter is documented inconsistently.
* Made the -x documentation and behavior consistent.
* Since this is also a new year, updated the copyright notices while here.
(closes issue ASTERISK-19094)
Reported by: Eugene
Patches:
issueA19094_correct_asterisk_option_x.patch (license #5674) patch uploaded by Walter Doekes (modified)
Tested by: Eugene
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ast_streamfile previously did unconditional seeking on files that broke
playback of formats that don't support that functionality. This patch avoids
the seek that was causing the problem. This regression was introduced in
r158062.
(closes issue ASTERISK-18994)
Patch-by: Timo Teras
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key press.
When receiving calls from a mobile phone into a DISA system on a connection with
significant interference, the reporter's Asterisk system would interpret DTMF incorrectly
and replicate digits received. This patch resolves that by increasing the number of
frames a mismatch has to be detected before assuming the DTMF is over by 1 frame and
adjusts dtmf_detect function to reset hits and misses only when an edge is detected.
(closes issue ASTERISK-17493)
Reported by: Alec Davis
Patches:
bug18904-refactor.diff.txt uploaded by Alec Davis (license 5546)
Review: https://reviewboard.asterisk.org/r/1130/
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When catching a signal, in no fork mode the console thread is identical to the thread
responsible for catching the signal and closing Asterisk, which requires it to first
dispense with the console thread. Prior to this patch, if these threads were identical,
upon receiving a killing signal, the thread will send an URG signal to itself, which
we also catch and then promptly do nothing with. Obviously this isn't useful behavior.
(closes issue ASTERISK-19127)
Reported By: Bryon Clark
Patches:
quit_on_signals.patch uploaded by Bryon Clark (license 6157)
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in CHANNEL()
(closes issue ASTERISK-18962)
reported by: Nir Simionovich
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Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
causes the loop to exit prematurely. This causes a variety of negative side
effects, depending on when the loop exits. This patch handles the frame by
essentially swallowing the frame in the local loop, as the current channel
drivers expect the RTP bridge to handle the frame, and, in the case of the
local bridge loop, no additional action is necessary.
(issue ASTERISK-19040)
(issue ASTERISK-19128)
(issue ASTERISK-17725)
(issue ASTERISK-18340)
(closes issue ASTERISK-19095)
Reported by: Stefan Schmidt
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1640/
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of duplicating that logic.
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This commit removes the V.21 preamble detection code previously added to the
generic DSP implementation in Asterisk, and instead enhances the res_fax module
to be able to utilize V.21 preamble detection functionality made available by
FAX technology modules. This commit also adds such support to res_fax_spandsp,
which uses the Spandsp modem tone detection code to do the V.21 preamble
detection.
There should be no functional change here, other than much more reliable V.21
preamble detection (and thus T.38 gateway initiation).
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frames, even after all of the hooks are detached. This patch short-cicuits us
out before we transcode unnecessarily.
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Adds two new options to SIP peers allowing them to specify features (dynamic or builtin)
to use when sending INFO/record requests. Recordonfeature activates whatever feature
is specified when recieving a record: on request while recordofffeature activates
whatever feature is specified when receiving a record: off request. Both of these
features can be disabled by setting the feature to an empty string.
(closes issue ASTERISK-16507)
Reported by: Jon Bright
Review: https://reviewboard.asterisk.org/r/1634/
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Chan_sip gives a dialog reference to the extension state callback and
assumes that when ast_extension_state_del() returns, the callback cannot
happen anymore. Chan_sip then reduces the dialog reference count
associated with the callback. Recent changes (ASTERISK-17760) have
resulted in the potential for the callback to happen after
ast_extension_state_del() has returned. For chan_sip, this could be very
bad because the dialog pointer could have already been destroyed.
* Added ast_extension_state_add_destroy() so chan_sip can account for the
sip_pvt reference given to the extension state callback when the extension
state callback is deleted.
* Fix pbx.c awkward statecbs handling in ast_extension_state_add_destroy()
and handle_statechange() now that the struct ast_state_cb has a destructor
to call.
* Ensure that ast_extension_state_add_destroy() will never return -1 or 0
for a successful registration.
* Fixed pbx.c statecbs_cmp() to compare the correct information. The
passed in value to compare is a change_cb function pointer not an object
pointer.
* Make pbx.c ast_merge_contexts_and_delete() not perform callbacks with
AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for
deadlocking when those locks are held during the callback.
* Removed unused lock declaration for the pbx.c store_hints list.
(closes issue ASTERISK-18844)
Reported by: rmudgett
Review: https://reviewboard.asterisk.org/r/1635/
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This patch adds initial testsuite event hooks so that ConfBridge tests
can be executed in the Asterisk TestSuite.
(issue ASTERISK-19059)
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(Closes issue ASTERISK-19056)
Reported by: Yuri
Patches:
348360.diff uploaded by Yuri (license #5242)
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* Add locking when a channel inherits variables and datastores in
__ast_request_and_dial() and ast_call_forward(). Note: The involved
channels are not active so there was minimal potential for problems.
* Remove calls to ast_set_callerid() in __ast_request_and_dial() and
ast_call_forward() because the set information is for the wrong direction.
* Don't use C++ keywords for variable names in ast_call_forward().
* Run the redirecting interception macro if defined when forwarding a call
in ast_call_forward(). Note: Currently will never execute because the
only callers that supply a calling channel supply a hungup or zombie
channel.
* Make feature_request_and_dial() put the transferee into autoservice when
it calls ast_call_forward() in case a redirection interception macro is
run. Note: Currently will never happen because the caller channel (Party
B) is always hungup at this time.
* Make feature_request_and_dial() ignore the AST_CONTROL_PROCEEDING frame
to silence a log message.
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(issue ASTERISK-18836)
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The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to
be called by different threads for the same channel. The channel driver
thread and the PBX thread running dialplan.
* Add lock protection around CDR API calls that access an ast_channel
pointer.
(closes issue ASTERISK-18836)
Reported by: gpluser
Review: https://reviewboard.asterisk.org/r/1628/
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PARKEDCALL variable tracks which parking lot the call was last parked in. This can be
used afterwards for flow control when returntoorigin is set to off. I went ahead and
documented both this and the existing variable set during timeout (PARKINGSLOT) in
the sample features.conf since there was no prior mention of variables being set during
timeout.
(closes issue ASTERISK-16239)
Reported By: Clod Patry
Patches:
M17503.diff uploaded by Clod Patry (license 5138)
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During testing, it was discovered that there were a number of side effects
introduced by r346391 and subsequent check-ins related to it (r346429,
r346617, and r346655). This included the /main/stdtime/ test 'hanging',
as well as the remote console option failing to receive the appropriate output
after a period of time.
I only backed out the changes to main/ and utils/, as this was adequate
to reverse the behavior experienced.
(issue ASTERISK-18974)
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* Simplify compare_char() and avoid potential sign extension issue.
* Fix infinite loop in add_exten_to_pattern_tree() handling of character
set escape handling.
* Added buffer overflow checks in add_exten_to_pattern_tree() character
set collection.
* Made ignore empty character sets.
* Added escape character handling to end-of-range character in character
sets. This has a slight change in behavior if the end-of-range character
is an escape character. You must now escape it.
* Fix potential sign extension issue when expanding character set ranges.
* Made remove duplicated characters from character sets. The duplicate
characters lower extension matching priority and prevent duplicate
extension detection.
* Fix escape character handling when the escape character is trying to
escape the end-of-string. We could have continued processing characters
after the end of the exten string. We could have added the previous
character to the pattern matching tree incorrectly.
(closes issue ASTERISK-18909)
Reported by: Luke-Jr
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When a bridge is broken, ast_bridge_call() might execute the h exten on
the calling channel. However, that channel may not have been the channel
that broke the bridge by hanging up. The channel executing the h exten
must be in a hung up state so things like AGI run in the correct mode.
* Make sure ast_bridge_call() marks the channel it is executing the h
exten on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as to
match the pbx.c main dialplan execution loop when it executes the h
exten.)
(closes issue ASTERISK-18811)
Reported by: David Hajek
Patches:
jira_asterisk_18811_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: David Hajek, rmudgett
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(closes issue ASTERISK-18958)
Reported by: Red
Patches:
jira_asterisk_18958_v1.8.patch (license #5621) patch uploaded by rmudgett
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This patch also makes astdb2sqlite3 and astcanary use the configured
directory instead of relying on $PATH.
(closes issue ASTERISK-18959)
Review: https://reviewboard.asterisk.org/r/1613/
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If waitExten specifies a music class to use with its music on hold option, it will use
CHANNEL(musicclass) instead if that channel variable has been set on the initiating
channel. This documents that behavior in the waitExten app so that this can be known
without checking the documentation of the code in function local_ast_moh_start.
(closes issue ASTERISK-18804)
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Prior to this patch, if labels with the same name were used for different priorities in
the same extension, the new label would be accepted, but it would be unusable since
attempts to reach that label would just go to the first one. Now pbx.c detects this,
generates a warning in logs, and culls the label before adding it to the dialplan.
(closes issue ASTERISK-18807)
Reported by: Kenneth Shumard
Patches:
pbx.c.patch uploaded by Kenneth Shumard (License 5077)
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The STUN socket must remain open between polls or the external address
seen by the STUN server is likely to change. However, if the STUN request
poll fails then the STUN server address needs to be re-resolved and the
STUN socket needs to be closed and reopened.
* Re-resolve the STUN server address and create a new socket if the STUN
request poll fails.
* Fix ast_stun_request() return value consistency.
* Fix ast_stun_request() to check the received packet for expected message
type and transaction ID.
* Fix ast_stun_request() to read packets until timeout or an associated
response packet is found. The stun_purge_socket() hack is no longer
required.
* Reduce ast_stun_request() error messages to debug output.
* No longer pass in the destination address to ast_stun_request() if the
socket is already bound or connected to the destination.
(closes issue ASTERISK-18327)
Reported by: Wolfram Joost
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1595/
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Cleaning up chan_sip/tcptls file descriptor closing.
This patch attempts to eliminate various possible instances of undefined behavior caused
by invoking close/fclose in situations where fclose may have already been issued on a
tcptls_session_instance and/or closing file descriptors that don't have a valid index
for fd (-1). Thanks for more than a little help from wdoekes.
(closes issue ASTERISK-18700)
Reported by: Erik Wallin
(issue ASTERISK-18345)
Reported by: Stephane Cazelas
(issue ASTERISK-18342)
Reported by: Stephane Chazelas
Review: https://reviewboard.asterisk.org/r/1576/
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This patch attempts to eliminate various possible instances of undefined behavior caused
by invoking close/fclose in situations where fclose may have already been issued on a
tcptls_session_instance and/or closing file descriptors that don't have a valid index
for fd (-1). Thanks for more than a little help from wdoekes.
(closes issue ASTERISK-18700)
Reported by: Erik Wallin
(issue ASTERISK-18345)
Reported by: Stephane Cazelas
(issue ASTERISK-18342)
Reported by: Stephane Chazelas
Review: https://reviewboard.asterisk.org/r/1576/
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verbosity level.
Review: https://reviewboard.asterisk.org/r/1599
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https://origsvn.digium.com/svn/asterisk/branches/10
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r346349 | dvossel | 2011-11-28 18:00:11 -0600 (Mon, 28 Nov 2011) | 10 lines
Fixes memory leak in message API.
The ast_msg_get_var function did not properly decrement
the ref count of the var it retrieves. The way this is
implemented is a bit tricky, as we must decrement the var and then
return the var's value. As long as the documentation for the
function is followed, this will not result in a dangling pointer as
the ast_msg structure owns its own reference to the var while it
exists in the var container.
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The dnsmgr refresh would always get the first address found regardless of
the original address family requested. So if you asked for only IPv4
addresses originally, you might get an IPv6 address on refresh.
* Saved the original address family requested by ast_dnsmgr_lookup() to be
used when the address is refreshed.
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This provides the same functionality as #include however an asterisk module will
still load if the filename does not exist.
Review: https://reviewboard.asterisk.org/r/1476/
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Review: https://reviewboard.asterisk.org/r/1593/
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In 1.8 and previous versions, one could use any fullword portion of the key
name, including the full key, to obtain the record. Until this patch, this
did not work for the full key.
Closes issue ASTERISK-18886
Patch: by tilghman
Review: by twilson (http://pastebin.com/7rtu6bpk) on #asterisk-dev
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(closes issue ASTERISK-18857)
Reported by: David M
Patches:
mainpbx-trivial.patch (License #6326) patch uploaded by David M
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int blah = 1;
...
write(chan->alertpipe[1], &blah, new_frames * sizeof(blah)) !=
(new_frames * sizeof(blah)))
is only valid when new_frames == 1. Otherwise we start reading into adjacent
variables declared on the stack. The read end discards what is read, so the
values don't matter but it's not a good idea to read past where we want even
though new_frames is almost always 1 and should never be large. This patch is
basically taken out of kpfleming's eventfd branch, as he mentioned that he
remembered fixing it there when I talked to him about this issue.
Review: https://reviewboard.asterisk.org/r/1583/
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Kevin P Fleming suggested that r343157 should use __alignof__ instead
of sizeof. For most systems this won't be an issue, but better fix it
now while it's still fresh.
Review: https://reviewboard.asterisk.org/r/1573
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This patch fixes the format type check in ast_closestream and
filestream_destructor. Previously a comparison operator was used, but since
audio formats are no longer contiguous (and AST_FORMAT_AUDIO_MASK includes
formats that have a value greater than the video formats), a bitwise AND
operation is used instead. Duplicated code was also moved to filestream_close.
(closes issue ASTERISK-18682)
Reported by: Aldo Bedrij
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1580/
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The opaque_data was added and enclosed in single quotes, assuming it
would be only a single line. The rest of the lines were appended after
the closing quote.
(closes issue ASTERISK-18852)
Reported by: peep_ on IRC
Review: https://reviewboard.asterisk.org/r/1577
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Holding the channel lock while the CLI "core show channel" command is
executing can slow down the system. It could block the system if the
console output is halted or paused.
* Made capture the CLI "core show channel" output into a buffer to be
output after the channel is unlocked.
* Removed use of C++ keyword as a variable name. out renamed to obuf.
* Checked allocation of obuf for failure so will not crash.
(closes issue ASTERISK-18571)
Reported by: Pavel Troller
Tested by: rmudgett
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Attempting to call an extension which used Caller ID matching with a channel that
has an empty caller id string would result in a segmentation fault.
(closes issue ASTERISK-18392
Reported By: Ales Zelenik
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https://origsvn.digium.com/svn/asterisk/branches/10
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r344493 | dvossel | 2011-11-10 15:54:42 -0600 (Thu, 10 Nov 2011) | 12 lines
Fixes issue with ConfBridge participants hanging up during DTMF feature menu usage getting stuck in conference forever.
When a conference user enters the DTMF menu they are suspended from the
bridge while the channel is handed off to the DTMF feature code. If a
user entered this state and hungup, there existed a race condition where
the channel could not exit the conference because it was waiting on a
signal that would never arrive. This patch fixes that, because it would
stupid for me to talk about the problem and commit a patch for something else.
(closes issue ASTERISK-18829)
Reported by: zvision
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* Fixed a LOG_ERROR message referencing the config variable list v that
had previously been processed and became NULL.
* Added error return value set that was missing in an ast_append_ha()
error return path.
(closes issue ASTERISK-18743)
Reported by: Michele
Patches:
issueA18743-fix_dynamic_exclude_static_bad_host_log.patch (license #5674) patch uploaded by Walter Doekes
Tested by: Michele
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The udptl port range was defined as 4000-4999 in the udptl.conf.sample,
as 4500-4599 if you didn't have a config and 4500-4999 if your config
was broken. Default is now 4000-4999.
(closes issue ASTERISK-16250)
Reviewed by: Tilghman Lesher
Review: https://reviewboard.asterisk.org/r/1565
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command "core show codec"
AST-699
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