A bug in the pbx_lua goto detection was causing the dialplan to hangup
unexpectedly after confbridge exited if it had called lua dialplan code during
execution.
Patch-by: Timo Teras
Acked-by: Matt Nicholson
(closes issue ASTERISK-18976)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Even though we set the frame to the ast_null_frame and return that,
the caller of the frame hook may still need the frame. This now is
a bit more careful about when it frees the frame, i.e., only under
the same conditions that applied when we duplicated it in the first
place.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a table is created by some other application and the primary key is not
named "AcctId", cel/cel_sqlite3_custom.c will always try to create the
table and fail because it already exists.
* Change the SQL table query to not require AcctId as the primary key.
(closes issue ASTERISK-18963)
Reported by: socketpair
Patches:
fix.patch (license #6337) patch uploaded by socketpair
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349673 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When looking up a ConfBridge profile, the config parser would, if it
found a channel datastore on the channel requesting the bridge profile,
unlock the channel mutex twice. Since that's a little aggressive,
it now only unlocks it once.
(closes issue ASTERISK-19042)
Reported by: Matt Jordan
Tested by: Matt Jordan
Patches:
19042 uploaded by David Vossel (license 5628)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A frame that is translated via ast_translate is also duplicated via ast_frdup.
This will allocate a new frame on the heap, which needs to be free'd
at the appropriate time. This issue reporter used valgrind to find that this
occurred in res_fax's fax_gateway_framehook; a quick search through the code
showed that only place this was currently not handling the translatted frame
properly.
(closes issue ASTERISK-19133)
Reported by: Sylvain Rochet
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added information collection from the output of the utilities: top, free, uptime, ifconfig
Added information collection from the output of the Asterisk command 'dahdi show status'
Added option / flag '-n, --non-interactive'
Updated man page to reflect new option / flag '-n, --non-interactive'
Patch-by: John Bigelow (itzanger)
(closes issue AST-749)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349340 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, res_musiconhold existed at the same module priority level
as the timing sources that it depends on. This would cause a problem when
music on hold was reloaded, as the timing source could be changed after
res_musiconhold was processed. This patch adds a new module priority level,
AST_MODPRI_TIMING, that the various timing modules are now loaded at. This
now occurs before loading other resource modules, such that the timing source
is guaranteed to be set prior to resolving the timing source dependencies.
(closes issue ASTERISK-17474)
Reporter: Luke H
Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
Patches:
asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff uploaded by elguero (License #5026)
asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff uploaded by elguero (License #5026)
asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by elguero (License #5026)
Review: https://reviewboard.asterisk.org/r/1578/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r349044 | seanbright | 2011-12-23 12:25:01 -0500 (Fri, 23 Dec 2011) | 18 lines
In ChanSpy, don't create audiohooks that will never be used.
When ChanSpy is initialized it creates and attaches 3 audiohooks:
1) Read audio off of the channel that we are spying on
2) Write audio to the channel that we are spying on
3) Write audio to the channel that is bridged to the channel that we are
spying on.
The first is always necessary, but the others are used only when specific
options are passed to the ChanSpy application (B, d, w, and W to be specific).
When those flags are not passed, neither of those audiohooks are ever sent
frames, but we still try to process the hooks for each voice frame that we
recieve on the channel.
So in short - only create and attach audiohooks that we actually need.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@349045 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There were a number of issues in cel_pgsql's pgsql_log method:
* If either sql or sql2 could not be allocated, the method would return while
the pgsql_lock was still locked
* If the execution of the log statement succeeded, the sql and sql2 structs
were never free'd
* Reconnection successes were logged as ERRORs. In general, the severity of
several logging statements was reduced
(closes issue ASTERISK-18879)
Reported by: Niolas Bouliane
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1624/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds initial testsuite event hooks so that ConfBridge tests
can be executed in the Asterisk TestSuite.
(issue ASTERISK-19059)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
According to the RTP packetization documentation, and the maximum values
listed in AST_FORMAT_LIST, we should support values > that the signed
char array that ast_codec_pref makes available to store the value. All
places in the code treat the framing field as though it were an int
array instaead of a char array anyway, so this just fixes the type of
the array.
(closes issue ASTERISK-18876)
Review: https://reviewboard.asterisk.org/r/1639/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348845 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some ISDN switches complain or block the call if the RDNIS number is
empty.
* Made chan_iax2 not save a RDNIS number into the ast_channel if the
string is blank. This is what other channel drivers do.
(closes issue ASTERISK-17152)
Reported by: rmudgett
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Merged revisions 348735 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348736 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Support weak symbols on a platform specific basis. The Mac OS X (Darwin)
support must be isolated from the other platforms because it has caused
other platforms to crash. Several other platforms including Linux have
GCC versions that define the weak attribute. However, this attribute is
only setup for use in the code by Darwin.
(closes issue ASTERISK-18728)
Reported by: Ben Klang
Review: https://reviewboard.asterisk.org/r/1617/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348465 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ParkAndAnnounce tried to pass the CallerID to the announcing channel but
the ID was wiped out by the channel masquerade done when parking the call.
* Save the CallerID before parking the channel to pass it to the
announcing channel.
* Fixed a minor memory leak in ParkAndAnnounce.
* Updated some ParkAndAnnounce log messages.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously, app_originate could not originate a call into a non-8kHz conference
bridge as the formats for non-8kHz slin codecs were not applied to the created
channel. This patch adds all of the formats by default, such that if a created
channel has a codec that supports a higher sampling rate, a translation path
can be built between it and other channels.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348265 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The function QUEUE_MEMBER has two required parameters (queuename, option). It
was only checking for the presence of queuename. The patch checks for the
existence of the option parameter and provides better error logging when
invalid values are provided for the option parameter as well.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The addition of the Connected Line support changed how CallerID is passed
to outgoing calls. The FollowMe application was not updated to pass
CallerID to the outgoing calls.
* Fix FollowMe CallerID on outgoing calls.
* Restructured findmeexec() to fix several memory leaks and eliminate some
duplicated code.
* Made check the return value of create_followme_number(). Putting a NULL
into the numbers list is bad if create_followme_number() fails.
* Fixed a couple uses of ast_strdupa() inside loops.
* The changes to bridge_builtin_features.c fix a similar CallerID issue
with the bridging API attended and blind transfers. (Not used at this
time.)
(closes issue ASTERISK-17557)
Reported by: hamlet505a
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1612/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The function ast_srtp_protect used a common buffer for both SRTP and SRTCP
packets. Since this function can be called from multiple threads for the same
SRTP session (scheduler for SRTCP and channel for SRTP) it was possible for the
packets to become corrupted as the buffer was used by both threads
simultaneously.
This patch adds a separate buffer for SRTCP packets to avoid the problem.
(closes issue ASTERISK-18889, Reported/patch by Daniel Collins)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
r325483 caused a regression in Asterisk 10+ that would make Asterisk segfault when
attempting to set penalty on an interface without specifying a queue in the queue set
penalty CLI command. In addition, no attempt would be made whatsoever to perform the
penalty setting on all the queues in the core list with either the cli command or the
non-segfaulting ami equivalent. This patch fixes that and also makes an attempt to
document and rename some functions required by this command to better represent what
they actually do. Oh yeah, and the use of this command without specifying a specific
queue actually works now.
Review: https://reviewboard.asterisk.org/r/1609/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Terry Wilson
Kinsey Moore
Richard Mudgett
Matthew Jordan
Jonathan Rose
Sean Bright
Kevin P. Fleming
Leif Madsen
Matthew Nicholson
Stefan Schmidt
Walter Doekes