Upon reload the module unconditionally "unloaded" the module (freeing memory
and setting pointers to NULL) and then when attempting a "load" if the config
file had not changed then nothing would be reinitialized.
By moving the "unload" to occur conditionally (reload only) after an attempted
configuration load, but before module "loading" alleviates the issue. The module
now loads/unloads/reloads correctly.
(closes issue ASTERISK-22871)
Reported by: Matteo
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
dahdi show channels output slices the callerid (which is dnid copied over on
PRI channels). If the channel naming structures look like:
'DAHDI/i1/1408409XXXX-6'
then the output slices 1408409XXXX down to 1408409XXX. This patch just opens
it up to 15 chars so you can see the whole thing.
(closes issue ASTERISK-22918)
Reported by: outtolunc
Patches:
svn_chan_dahdi.c.format12_15.diff.txt uploaded by outtolunc (license 5198)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When calling REPLACE() with an empty replace-char argument, strcpy
is used to overwrite the the matching <find-char>. However as the
src and dest arguments to strcpy must not overlap, it causes other
parts of the string to be overwritten with adjacent characters and
the result is mangled. Patch replaces call to strcpy with memmove
and adds a test suite case for REPLACE.
(closes issue ASTERISK-22910)
Reported by: Gareth Palmer
Review: https://reviewboard.asterisk.org/r/3083/
Patches:
func_strings.patch uploaded by Gareth Palmer (license 5169)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A deadlock can happen between a thread unloading or reloading the cel_pgsql
module and the core_event_dispatcher taskprocessor thread. Description of
what is happening:
Thread 1 (for example, a netconsole thread):
a "module reload cel_pgsql" is launched
the thread enter the "my_unload_module" function (cel_pgsql.c)
the thread acquire the write lock on psql_columns
the thread enter the "ast_event_unsubscribe" function (event.c)
the thread try to acquire the write lock on ast_event_subs[sub->type]
Thread 2 (core_event_dispatcher taskprocessor thread):
the taskprocessor pop a CEL event
the thread enter the "handle_event" function (event.c)
the thread acquire the read lock on ast_event_subs[sub->type]
the thread callback the "pgsql_log" function (cel_pgsql.c), since it's a subscriber of CEL events
the thread try to acquire a read lock on psql_columns
(closes issue ASTERISK-22854)
Reported by: Etienne Lessard
Patches:
cel_pgsql_fix_deadlock_event.patch uploaded by hexanol (license 6394)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
'core show channeltypes' type column is being sliced, resulting in incomplete
type names.
(closes issue ASTERISK-22919)
Reported by: outtolunc
Patches:
svn_channel.c.format_15.diff.txt uploaded by outtolunc (license 5198)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In fax_detect_framehook() a null pointer reference can occur where a
voice frame is processed but no dsp is attached to the fax detection
structure. The code block that rejects frames that detection cannot
be processed on is checking for dsp but falls through when it should
instead return, as this change implements.
(closes issue ASTERISK-22942)
Reported by: adomjan
Review: https://reviewboard.asterisk.org/r/3076/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When Asterisk is shut down, the astdb_atexit() function releases
(finalize) the previously initiated (prepared) SQL statements in
sqlite3. Another thread making a subsequent request can cause a
crash in sqlite3. This patch eliminates that issue by resetting
the statement pointer after it is released/cleared. The sqlite3
code detects the null pointer, and aborts the operation cleanly.
(closes issue AST-1265)
Reported by: Alexander Hömig
(closes issue ASTERISK-22350)
Reported by: Birger "WIMPy" Harzenetter
Review: https://reviewboard.asterisk.org/r/3078/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Introduce new 'stopped' state for gk client and restart gk client
on failures
Remove ooh323 stack command lock as it is not need now.
(closes issue ASTERISK-21960)
Reported by: Dmitry Melekhov
Patches:
ASTERISK-21960.patch
ASTERISK-21960-stacklockup-2.patch
Tested by: Dmitry Melekhov
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@404318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch allows individual dialplan functions to be marked as
'dangerous', to inhibit their execution from external sources.
A 'dangerous' function is one which results in a privilege escalation.
For example, if one were to read the channel variable SHELL(rm -rf /)
Bad Things(TM) could happen; even if the external source has only read
permissions.
Execution from external sources may be enabled by setting
'live_dangerously' to 'yes' in the [options] section of asterisk.conf.
Although doing so is not recommended.
(closes issue ASTERISK-22905)
Review: http://reviewboard.digium.internal/r/432/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@403917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
During dialplan execution in pbx_extension_helper(), the contexts global
read lock prevents link list corruption, but was released with a pointer
to the ast_exten and data later used in variable substitution. Instead,
this patch removes pbx_substitute_variables() and locates a copy of the
ast_exten data on the stack before releasing the lock, where ast_exten
could get free'd by another thread performing a module reload.
(issue AST-1179)
Reported by: Thomas Arimont
(issue AST-1246)
Reported by: Alexander Hömig
Review: https://reviewboard.asterisk.org/r/3055/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@403863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch prevents an infinite loop overwriting memory when
a message is received into the unpacksms16() function, where
the length of the message is an odd number of bytes.
(closes issue ASTERISK-22590)
Reported by: Jan Juergens
Tested by: Jan Juergens
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@403855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, res_fax_spandsp was conservative with how it initialized
the spandsp T.38 context. It would only initialize it if the driver thought
the current state was a T.38 fax. While this works fine in nominal situations,
in certain off nominal situations, res_fax_spandsp can believe that a T.38
fax will not occur when in fact one has started. In particular, this was
discovered when res_fax would fall back to audio after timing out on a T.38
upgrade. The SIP channel driver would continue to retry the re-INVITE and -
if the remote end responded after res_fax timed out with a 200 OK - a T.38
frame would be delivered to the res_fax stack when it no longer expected it.
As it turns out, there does not appear to be any downside to always
initializing the T.38 context, other than the actual memory allocation.
Since that avoids this off nominal situation (and others which are equally
likely hard to predict), this is the safest way to avoid this problem.
Much thanks to Torrey as well for providing a scenario that reproduces this
issue.
(closes issue ASTERISK-21242)
Reported by: Ashley Winters
Tested by: Torrey Searle
patches:
always-init-t38.patch uploaded by awinters (License 6477)
A_PARTY.xml uploaded by tsearle (License 5334)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@403450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When translating from one format to another it is possible
to inform the translation function that the source frame should
be freed. This was previously done immediately but shortly
afterwards the frame that was freed was accessed and used again.
This change moves code around a bit so that the frame is now
freed after it has been completely used.
(closes issue ASTERISK-22788)
Reported by: Corey Farrell
Patches:
translate-access-after-free-11up.patch uploaded by coreyfarrell (license 5909)
translate-access-after-free-1.8.patch uploaded by coreyfarrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@403015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk will sometimes core dump during caller id read on analog
channels due to a negative return value from the read() in
my_get_callerid that slips through as a negative length argument to
callerid_feed() if the errno returned by DAHDI is ELAST. This change
ensures that the negative return is treated properly even when it is
ELAST.
(closes issue ASTERISK-22746)
Reported by: Michael Walton
Patches:
chan_dahdi_cid_crash_fix.r401410.patch uploaded by Michael Walton (License 6502)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In the current app_queue code from 1.8 up to trunk the upper and lower
penalties can be set to 0 but the value is interpreted to be disabled
instead of actually setting limits. This is especially evident if min
and max limits are set to 0 and members with penalties of 0 and 1 are
in the queue since the member with penalty 1 will still receive calls.
This patch adjusts the special disabled value to be INT_MAX instead of
0.
(closes issue ASTERISK-20862)
Review: https://reviewboard.asterisk.org/r/2995/
Reported by: Schmooze Com
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402646 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For outbound register requests the tag on the From line was
updated every 20 seconds prior to a successful registration
and also once for each registration renewal. That behavior
can possibly cause the registration to be denied because of
the different tag, and is not aligned with the intention of
RFC 3261 8.1.3.5 "... request constitutes a new transaction
and SHOULD have the same value of the Call-ID, To, and From
of the previous request...". This updates chan_sip to have
a field to keep the local tag in the registration structure
and use that tag for registration requests where the callid
is also unchanged.
(closes issue ASTERISK-12117)
Reported by: Pawel Pierscionek
Review: https://reviewboard.asterisk.org/r/2988/
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........
chan_sip: notify dialog info ignores presentation indicator in callerid
The presentation indicator in a callerid (e.g. set by dialplan function
Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog Info Notifies
are generated during extension monitoring. Added a check to make sure the
name and/or number presentations on the callee (remote identity) are set to
allow. If they are restricted then "anonymous" is used instead.
(closes issue AST-1175)
Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/2976/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The presentation indicator in a callerid (e.g. set by dialplan function
Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog Info Notifies
are generated during extension monitoring. Added a check to make sure the
name and/or number presentations on the callee (remote identity) are set to
allow. If they are restricted then "anonymous" is used instead.
(closes issue AST-1175)
Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/2976/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The system overrides the user muting requests when MOH is playing or a
waitmarked user is waiting for a marked user to join. System muting
overrides interfere with what the user may wish the muting to be when the
system override ends.
* User muting requests are now independent of the system muting overrides.
The effective muting is now the logical or of the user request and system
override.
* Added a Muted column to the CLI "confbridge list <conference>" command.
* Added a Muted header to the AMI ConfbridgeList action ConfbridgeList
event.
(closes issue AST-1102)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2960/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ConfBridge allows custom DTMF menus to be created in the confbridge.conf
file by assigning a DTMF key sequence to a sequence of actions as follows:
DTMF-sequence = action,action...
Unfortunately, the normal config file processing code interprets an
initial '#' character as starting a directive such as #include.
* Add the ability to escape the first non-blank character in a config line
so the '#' character can be used without triggering the directive
processing code.
(closes issue AFS-2)
(closes issue ASTERISK-22478)
Reported by: Nicolas Tanski
Patches:
jira_asterisk_22478_v11.patch (license #5621) patch uploaded by rmudgett (modified)
Review: https://reviewboard.asterisk.org/r/2969/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This corrects one-way audio between Asterisk and Chrome/jssip as a
result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX
ICE candidates. This also exposes an ICE component enumeration to
extract further details from candidates.
(closes issue ASTERISK-21383)
Reported by: Shaun Clark
Review: https://reviewboard.asterisk.org/r/2967/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For awhile now, we've noticed continuous integration builds hanging on CentOS 6
64-bit build agents. After resolving a number of problems with symbols, strange
locks, and other shenanigans, the problem has persisted. In all cases, gdb
shows the Asterisk process stuck in loader.c on one of the infinite while loops
that calls dlclose repeatedly until success.
The documentation of dlclose states that it returns 0 on success; any other
value on error. It does not state that repeatedly calling it will eventually
clear those errors. Most likely, the repeated calls to dlclose was to force a
close by exhausting the references on the library; however, that will never
succeed if:
(a) There is some fundamental error at work in the loaded library that
precludes unloading it
(b) Some other loaded module is referencing a symbol in the currently loaded
module
This results in Asterisk sitting forever.
Since we have matching pairs of dlopen/dlclose, this patch opts to only call
dlclose once, and log out as an ERROR if dlclose fails to return success. If
nothing else, this might help to determine why on the CentOS 6 64-bit build agent
things are not closing successfully.
Review: https://reviewboard.asterisk.org/r/2970
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The new sound packages relate to issues: ASTERISK-22544, ASTERISK-22411, ASTERISK-21413, ASTERISK-20782
Modified sounds/Makefile for the new sound versions and to account for the new en_GB language set.
(issue ASTERISK-22659)
(closes issue ASTERISK-22659)
(closes issue ASTERISK-22411)
(closes issue ASTERISK-22544)
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Debug messages aren't free. Even when the debug level is sufficiently low such
that the messages are never evaluated, there is a cost to having to parse
Asterisk logs that contain debug messages that (a) fail to convey sufficient
information or (b) occur so frequently as to be next to meaningless. Based on
having to stare at lots of DEBUG messages, this patch makes the following
changes:
* channel.c: When copying variables from a parent channel to a child channel,
specify the channels involved. Do not log anything for a variable that is not
inherited; the fact that it doesn't have an _ or __ already signifies that it
won't be inherited.
* pbx.c: Specify what function evaluation has occurred that created the result.
* translate.c: Bump up the translator path messages to 10. I've never once had
to use these debug messages, and for each format that is registered (on
startup) and unregistered (on shutdown) the entire f^2 matrix is logged out.
For short tests in the Asterisk Test Suite, this should make finding the
actual test much easier.
* xmldoc.c: The debug message that 'blah' is not found in the tree is expected.
Often, description elements - which are not required - are not provided.
This debug message adds no additional value, as it is not indicative of an
error or helpful in debugging which element did not contain a 'blah' element
as a child. If an element is supposed to contain a child element, then that
XML tree should have failed validation in the first place.
Review: https://reviewboard.asterisk.org/r/2966/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While looking at ASTERISK-22236, Walter Doekes pointed out that when running
"sip show peers", the setting being displayed can be confusing. The display of
"N" used to mean NAT (i.e. yes). The NAT setting has gone through many
different changes resulting in the display of different characters to try and
convey what the current setting is for 'Forcerport' (A for Auto and Forcerport
is currently on, a for Auto but Forcerport is off, Y for yes, and N for no).
During the initial code review to try and clarify these settings (especially
since "N" no longer meant what it used to mean in prior versions of Asterisk),
Mark Michelson suggested using the full space available to display the settings
which helped to make the settings very clear. That was a great suggestion.
Therefore, this patch does the following:
* The column for 'Forcerport' now will show: Auto (Yes), Auto (No), Yes, or No.
* A column for the 'Comedia' setting has been added. It too will display the
setting in a non-cryptic way: Auto (Yes), Auto (No), Yes, or No.
* UPGRADE.txt has been updated to document this change.
(closes issue ASTERISK-22728)
Reported by: Walter Doekes
Tested by: Michael L. Young
Patches:
asterisk-forcerport-display-clarification_v3.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2941
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In function ast_rtp_instance_early _bridge_make_compatible the
use of instance 0/1 as arguments doesn't clearly communicate a
direction that the copying of payloads from the source channel
to the destination channel will occur, making it more probable
to have the arguments to ast_rtp_codecs_payloads_copy() put in
the reverse order. This patch renames the arguments with _dst
and _src suffixes and corrects the copy direction.
(closes issue ASTERISK-21464)
Reported by: Kevin Stewart
Review: https://reviewboard.asterisk.org/r/2894/
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Merged revisions 402000 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Test shows rtpmap:119 being copied per this change, but is not in sip invite
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes a bug where a zero length callerid match adjacent to a no
match callerid extension entry would be deleted together, which then
resulted in hashtable references to free'd memory. A third state of
the matchcid value has been added to indicate match to any extension
which allows enforcing comparison of matchcid on/off without errors.
(closes issue AST-1235)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/2930/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401960 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adapts the behaviour of avpf to only impact the format of outgoing calls. For
inbound calls, both AVP and AVPF calls will be accepted regardless of the value
of avpf in the configuration.
(closes issue ASTERISK-22005)
Reported by: Torrey Searle
Patches:
optional_avpf_trunk.patch uploaded by tsearle (license 5334)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If one specified a verbose level within a logging facility in
logger.conf then any component after it was ignored. Fixed so
all values are correctly read.
(closes issue ASTERISK-22456)
Reported by: Kevin Harwell
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Final set of patches in a series of memory leak/cleanup patches by Corey Farrell
(closes issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
main-utils-1.8.patch uploaded by coreyfarrell (license 5909)
main-utils-11.patch uploaded by coreyfarrell (license 5909)
main-utils-12up.patch uploaded by coreyfarrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also covers ast_app_parse_timelen-fail-zero-length.patch, but the patch was
replaced with one of my own.
(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
chan_dahdi-cleanup_push.patch uploaded by coreyfarrell (license 5909)
clicompat-r2.patch uploaded by coreyfarrell (license 5909)
codecs-ilbc-doCPLC.patch uploaded by coreyfarrell (license 5909)
data-cleanup-test-registration.patch uploaded by coreyfarrell (license 5909)
main-asterisk-kill-listener.patch uploaded by coreyfarrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
