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This fixes three main issues
* Change asprintf() uses to ast_asprintf() so that it
pairs properly with ast_free() and no longer causes
MALLOC_DEBUG to freak out.
* When ast_asprintf() fails, set the pointer NULL if
it will be referenced later.
* Fix some memory leaks that were spotted while taking
care of the first two points.
(Closes issue ASTERISK-20135)
reported by Richard Mudgett
Review: https://reviewboard.asterisk.org/r/2071
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This adds HANGUPCAUSE information to called channels so that hangup
handlers can, in conjunction with predial dialplan execution, access
the hangupcause information when the dialed channel hangs up on a
one-to-one basis instead of a many-to-one basis as with HANGUPCAUSE
usage on the caller channel.
Review: https://reviewboard.asterisk.org/r/2069/
(closes issue ASTERISK-20198)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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The flash-hook the bridged peer feature now correctly determines if the
bridged peer is another chan_dahdi channel, that it is an analog channel,
and that it has the correct signaling for an FXO port. It now also
flash-hooks the correct channel.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch adds named calledgroups/pickupgroups to Asterisk. Named groups are
implemented in parallel to the existing numbered callgroup/pickupgroup
implementation. However, unlike the existing implementation, which is limited
to a maximum of 64 defined groups, the number of defined groups allowed for
named callgroups/pickupgroups is effectively unlimited.
Named groups are configured with the keywords "namedcallgroup" and
"namedpickupgroup". This corresponds to the numbered group definitions of
"callgroup" and "pickupgroup". Note that as the implementation of named groups
coexists with the existing numbered implementation, a defined named group of
"4" does not equate to numbered group 4.
Support for the named groups has been added to the SIP, DAHDI, and mISDN channel
drivers.
Review: https://reviewboard.asterisk.org/r/2043
Uploaded by:
Guenther Kelleter(license #6372)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This replaces all calls to alloca() with ast_alloca() which calls gcc's
__builtin_alloca() to avoid BSD semantics and removes all NULL checks
on memory allocated via ast_alloca() and ast_strdupa().
(closes issue ASTERISK-20125)
Review: https://reviewboard.asterisk.org/r/2032/
Patch-by: Walter Doekes (wdoekes)
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The HANGUPCAUSE hash (trunk only) meant to replace SIP_CAUSE has now
been replaced with the HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan
functions to better facilitate access to the AST_CAUSE translations
for technology-specific cause codes. The HangupCauseClear application
has also been added to remove this data from the channel.
(closes issue SWP-4738)
Review: https://reviewboard.asterisk.org/r/2025/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Make starting analog_ss_thread() or __analog_ss_thread() failure paths
hangup the channel.
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This patch adds some basic documentation for a number of modules. This
includes core source files in Asterisk (those in main), as well as
chan_agent, chan_dahdi, chan_local, sig_analog, and sig_pri. The DTD
has also been updated to allow referencing of AMI commands.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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It is not necessary to generate information cause code frames on every
protocol event that occurs. This removes all the instances where the
frame was not conveying a cause code and was instead just conveying a
protocol-specific message. This also corrects the generation of the
message associated with disconnects for MFC/R2 to use the MFC/R2
specific text for the disconnect cause.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Adds call ID logging changes to specific channel drivers that weren't handled
handled in phase II of Call ID Logging. Also covers logging for threads for
threads created by systems that may be involved with many different calls.
Extra special thanks to Richard for rigorous review of chan_dahdi and its
various signalling modules.
review: https://reviewboard.asterisk.org/r/1927/
review: https://reviewboard.asterisk.org/r/1950/
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This adds a minimal implementation of the "Who Hung Up?" Asterisk 11
work to chan_dahdi.c for the MFC/R2 DAHDI subtech. Given the way that
OpenR2 interfaces with chan_dahdi, it is much harder to expose the
type of protocol information that is available in PRI, SS7, or other
channel technologies.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Channel drivers that allow native bridging need to handle
AST_CONTROL_PVT_CAUSE_CODE frames and previously did not handle them
properly, usually breaking out of the native bridge. This change
corrects that behavior and exposes the available cause code information
to the dialplan while native bridges are in place. This required
exposing the HANGUPCAUSE hash setter outside of channel.c, so
additional documentation has been added.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Calling ast_set_hangupsource() with the channel lock held can result in a
deadlock because the function also locks the bridged channel.
(issue ASTERISK-19537)
(closes issue AST-891)
Reported by: Guenther Kelleter
Tested by: Guenther Kelleter
(closes issue ASTERISK-19801)
Reported by: Alec Davis
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Most of these were just saving returned values without using them and
in some cases the variable being saved to could be removed as well.
(issue ASTERISK-19672)
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A deadlock can occur when a POTS phone tries to flash hook to originate a
second call for 3-way or transfer. If another process is scanning the
channels container when the POTS line flash hooks then a deadlock will
occur.
* Release the channel and private locks when creating a new channel as a
result of a flash hook.
(closes issue ASTERISK-19842)
Reported by: rmudgett
Tested by: rmudgett
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This addresses core findings 4 and 6.
Moises Silva helped me by stating that a break could be
safely added to the case where it is added in chan_dahdi.c
In say.c, I have added a comment indicating that static analysis
complains but that it is currently unknown if this is correct.
This fixes all core findings of this type.
(closes issue ASTERISK-19662)
reported by Matthew Jordan
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This patch adds to what was fixed in r366880. Specifically, it addresses the
following:
* chan_sip: dispose of an allocated frame in off nominal code paths in
sip_rtp_read
* func_odbc: when disposing of an allocated resultset, ensure that any rows
that were appended to that resultset are also disposed of
* cli: free the created return string buffer in another off nominal code
path
* chan_dahdi: free a frame that was allocated by the dsp layer if we choose
not to process that frame
(issue ASTERISK-19665)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1922/
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This patch addresses a number of memory leaks in a variety of modules that were
found by a static analysis tool. A brief summary of the changes:
* app_minivm: free ast_str objects on off nominal paths
* app_page: free the ast_dial object if the requested channel technology
cannot be appended to the dialing structure
* app_queue: if a penalty rule failed to match any existing rule list
names, the created rule would not be inserted and its memory
would be leaked
* app_read: dispose of the created silence detector in the presence of
off nominal circumstances
* app_voicemail: dispose of an allocated unique ID field for MWI event
un-subscribe requests in off nominal paths; dispose of
configuration objects when using the secret.conf option
* chan_dahdi: dispose of the allocated frame produced by ast_dsp_process
* chan_iax2: properly unref peer in CLI command "iax2 unregister"
* chan_sip: dispose of the allocated frame produced by sip_rtp_read's
call of ast_dsp_process; free memory in parse unit tests
* func_dialgroup: properly deref ao2 object grhead in nominal path of
dialgroup_read
* func_odbc: free resultset in off nominal paths of odbc_read
* cli: free match_list in off nominal paths of CLI match completion
* config: free comment_buffer/list_buffer when configuration file load
is unchanged; free the same buffers any time they were
created and config files were processed
* data: free XML nodes in various places
* enum: free context buffer in off nominal paths
* features: free ast_call_feature in off nominal paths of applicationmap
config processing
* netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct
that is allocated by the method. Failures in
ast_sockaddr_resolve could result in the users of the method
not knowing whether or not the buffer was allocated. The
method will now not allocate the ast_sockaddr struct if it
will return failure.
* pbx: cleanup hash table traversals in off nominal paths; free
ignore pattern buffer if it already exists for the specified
context
* xmldoc: cleanup various nodes when we no longer need them
* main/editline: various cleanup of pointers not being freed before being
assigned to other memory, cleanup along off nominal paths
* menuselect/mxml: cleanup of value buffer for an attribute when that attribute
did not specify a value
* res_calendar*: responses are allocated via the various *_request method
returns and should not be allocated in the various
write_event methods; ensure attendee buffer is freed if no
data exists in the parsed node; ensure that calendar objects
are de-ref'd appropriately
* res_jabber: free buffer in off nominal path
* res_musiconhold: close the DIR* object in off nominal paths
* res_rtp_asterisk: if we run out of ports, close the rtp socket object and free
the rtp object
* res_srtp: if we fail to create the session in libsrtp, destroy the
temporary ast_srtp object
(issue ASTERISK-19665)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1922
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This patch fixes two problems pointed out by a static analysis tool.
* In chan_dahdi, when an event is handled the index of the sub channel is first
obtained. In very off nominal cases, the method that determines the index
can return a negative value. In the event handling code, whether or not
the index returned is valid was being checked after that value was used to
index into an array. This patch makes it so the value is checked before
any indexing is done.
* In res_calendar_ews, sizeof was being passed a pointer instead of the struct to
determine the amount of memory to allocate.
(issue ASTERISK-19651)
Reported by: Matt Jordan
(closes issue ASTERISK-19671)
Reported by: Matt Jordan
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This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20,
22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111,
and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped
were either extended/deprecated or in areas of code that shouldn't be
disturbed.
(Closes issue ASTERISK-19650)
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disconnecting the call.
Some switches may not handle the call-deflection/call-rerouting message if
the call is disconnected too soon after being sent. Asteisk was not
waiting for any reply before disconnecting the call.
* Added a 5 second delay before disconnecting the call to wait for a
potential response if the peer does not disconnect first.
(closes issue ASTERISK-19708)
Reported by: mehdi Shirazi
Patches:
jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett
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Some ISDN switches occasionally fail to send a RESTART ACKNOWLEDGE in
response to a RESTART request.
* Made the second SETUP received after sending a RESTART request clear the
channel resetting state as if the peer had sent the expected RESTART
ACKNOWLEDGE before continuing to process the SETUP. The peer may not be
sending the expected RESTART ACKNOWLEDGE.
(issue ASTERISK-19608)
(issue AST-844)
(issue AST-815)
Patches:
jira_ast_815_v1.8.patch (license #5621) patch uploaded by rmudgett (modified)
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The CHANNEL_DEADLOCK_AVOIDANCE() feature of preserving where the channel
lock was originally obtained is overkill where ast_channel_lock_both() was
inlined.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Several telcos bring the BRI PTMP layer 1 down when the line is idle.
When layer 1 goes down, Asterisk cannot make outgoing calls. Incoming
calls could fail as well because the alarm processing is handled by a
different code path than the Q.931 messages.
* Add the layer1_presence configuration option to ignore layer 1 alarms
when the telco brings layer 1 down. This option can be configured by span
while the similar DAHDI driver teignorered=1 option is system wide. This
option unlike layer2_persistence does not require libpri v1.4.13 or newer.
Related to JIRA AST-598
JIRA ABE-2845
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Patch by: junky
Review: https://reviewboard.asterisk.org/r/1743/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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In chan_agent, while handling a channel indicate, the agent channel driver
must obtain a lock on both the agent channel, as well as the channel the
agent channel is using. To do so, it attempts to lock the other channel
first, then unlock the agent channel which is locked prior to entry into
the indicate handler. If this unlock fails with a negative return value,
which can occur if the object passed to agent_indicate is an invalid ao2
object or is NULL, the return value is passed directly to strerror, which
can only accept positive integer values.
In chan_dahdi, the return value of dahdi_get_index is used to directly
index into the sub-channel array. If dahd_get_index returns a negative
value, it would use that value to index into the array, which could cause
an invalid memory access. If dahdi_get_index returns a negative number,
we now default to SUB_REAL.
(issue ASTERISK-19655)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1863/
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In the MWI processing loop, when a valid event occurs the temporary caller ID
information is deallocated. If a new DAHDI channel is successfully created,
the event is passed up to the analog_ss_thread without error and the loop
exits. If, however, the DAHDI channel is not created, then the caller ID
struct has been free'd, and the gains reset to their previous level. This
will almost certainly cause an invalid access to the free'd memory, either
in subsequent calls to callerid_free or calls to callerid_feed.
* Rework the -r361705 patch to better manage the cs and mtd allocated
resources.
* Fixed use of mwimonitoractive flag to be correct if the mwi_thread()
fails to start.
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In the MWI processing loop, when a valid event occurs the temporary caller ID
information is deallocated. If a new DAHDI channel is successfully created,
the event is passed up to the analog_ss_thread without error and the loop
exits. If, however, the DAHDI channel is not created, then the caller ID
struct has been free'd, and the gains reset to their previous level. This
will almost certainly cause an invalid access to the free'd memory, either
in subsequent calls to callerid_free or calls to callerid_feed.
This patch makes it so that we only free the caller ID structure if a
DAHDI channel is successfully created, and we bump the gains back up
if we fail to make a DAHDI channel.
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Added new chan_dahdi.conf colp_send option parameter to block connected
line updates per span.
(closes issue ASTERISK-17025)
Reported by: Michael Smith
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Review: https://reviewboard.asterisk.org/r/1786/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* Fix referencing the wrong variable in chan_dahdi.c:my_set_cadence().
Thanks to Sean Bright for compiling with -Wshadow and finding this bug.
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For analog lines, enables Asterisk to use dialtone detection per channel
if an incoming call was hung up before it was answered. If dialtone is
detected, the call is hung up.
no: Disabled. (Default)
yes: Look for dialtone for 10000 ms after answer.
<number>: Look for dialtone for the specified number of ms after answer.
always: Look for dialtone for the entire call. Dialtone may return
if the far end hangs up first.
dialtone_detect=yes
dialtone_detect=5000
dialtone_detect=always
(closes issue ASTERISK-19316)
Reported by: Jeremy Pepper
Patch by: Jeremy Pepper
Tested by: rmudgett,Jeremy Pepper
Review: https://reviewboard.asterisk.org/r/1737/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Outgoing SS7 calls fail to detect incoming DTMF so any bridged channel
that requires out-of-band DTMF will not work.
* Added sig_ss7_open_media() calls at appropriate places in sig_ss7.c.
The new call converts conditionaled out unconverted code and shows that
the code really did something useful.
* Improved some chan_dahdi DTMF debug messages to help track DTMF
handling.
(closes issue ASTERISK-19312)
Reported by: Igor Nikolaev
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s/ast_channel_fd_set/ast_channel_internal_fd_set/g
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Review: https://reviewboard.asterisk.org/r/1784/
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Review: https://reviewboard.asterisk.org/r/1773/
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Review: https://reviewboard.asterisk.org/r/1770/
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Review: https://reviewboard.asterisk.org/r/1753/
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'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag.
sig_pri and sig_ss7 were keeping 'outgoing' flag insync.
Now provides a callback for all the low level sig_XXX modules.
(issue ASTERISK-19316)
alecdavis (license 585)
Reported by: Jeremy Pepper
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/1747/
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Review: https://reviewboard.asterisk.org/r/1733/
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needed.
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Patch by: Clod Patry
Review: https://reviewboard.asterisk.org/r/1651
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