XML encoding in chan_sip is accomplished by naively building the XML
directly from strings. While this usually works, it fails to take into
account escaping the reserved characters in XML.
This patch adds an 'ast_xml_escape' function, which works similarly to
'ast_uri_encode'. This is used to properly escape the local_display
attribute in XML formatted NOTIFY messages.
Several things to note:
* The Right Thing(TM) to do would probably be to replace the
ast_build_string stuff with building an ast_xml_doc. That's a much
bigger change, and out of scope for the original ticket, so I
refrained myself.
* It is with great sadness that I wrote my own ast_xml_escape
function. There's one in libxml2, but it's knee-deep in
libxml2-ness, and not easily used to one-off escape a
string.
* I only escaped the string we know is causing problems
(local_display). At least some of the other strings are
URI-encoded, which should be XML safe. Rather than figuring out
what's safe and escaping what's not, it would be much cleaner to
simply build an ast_xml_doc for the messages and let the XML
library do the XML escaping. Like I said, that's out of scope.
(closes issue ABE-2902)
Reported by: Guenther Kelleter
Tested by: Guenther Kelleter
Review: http://reviewboard.digium.internal/r/365/
........
Merged revision 378919 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The AMI redirect action can fail to redirect two channels that are bridged
together. There is a race between the AMI thread redirecting the two
channels and the bridge thread noticing that a channel is hungup from the
redirects.
* Made the bridge wait for both channels to be redirected before exiting.
* Made the AMI redirect check that all required headers are present before
proceeding with the redirection.
* Made the AMI redirect require that any supplied ExtraChannel exist
before proceeding. Previously the code fell back to a single channel
redirect operation.
(closes issue ASTERISK-18975)
Reported by: Ben Klang
(closes issue ASTERISK-19948)
Reported by: Brent Dalgleish
Patches:
jira_asterisk_19948_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Thomas Sevestre, Deepak Lohani, Kayode
Review: https://reviewboard.asterisk.org/r/2243/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378356 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk maintains an internal cache for devices in the event subsystem. The
device state cache holds the state of each device known to Asterisk, such that
consumers of device state information can query for the last known state for
a particular device, even if it is not part of an active call. The concept of
a device in Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls are allowed
in Asterisk. A device was automatically created and stored in the cache for
each anonymous call that occurred; this was possible in the SIP and IAX2
channel drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
are never removed from the system, allowing anonymous calls to potentially
exhaust a system's resources.
This patch changes the event cache subsystem and device state management to
no longer cache devices that are not associated with a physical entity.
(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
patches:
event-cachability-3.diff uploaded by jcolp (license 5000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The RTP engine public function that gets the available formats expects a
format_t to be returned; however when calling into an RTP instance's
callback to get the available formats, the callback returned an int.
This never was noticed in Asterisk because the two RTP engines included
do not provide an available_formats callback.
This introduces an API change, and the proposal for this change was brought
up on the Asterisk developers mailing list [1]. There was no public objection
to this change, so it is now being put in.
(closes AST-1054)
reported by Doug Bailey
[1] http://lists.digium.com/pipermail/asterisk-dev/2012-December/058058.html
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Cleanup time zones on exit.
* Make exit clean/unclean report consistent for AMI and CLI in
really_quit().
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
core-cleanup-1_8-10.patch (license #5909) patch uploaded by Corey Farrell
core-cleanup-11-trunk.patch (license #5909) patch uploaded by Corey Farrell
Modified
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@377135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Similar to the patch that moved the fork earlier in the startup sequence to
prevent mutex errors in the recursive mutex surrounding the read/write thread
registration lock, this patch re-initializes the logmsgs mutex. Part of the
start up sequence before forking the process into the background includes
reading asterisk.conf; this has to occur prior to the call to daemon in order
to read startup parameters. When reading in a conf file, log statements can
be generated. Since this can't be avoided, the mutex instead is
re-initialized to ensure a reset of any thread tracking information.
This patch also includes some additional debugging to catch errors when
locking or unlocking the recursive mutex that surrounds locks when the
DEBUG_THREADS build option is enabled. DO_CRASH or THREAD_CRASH will
cause an abort() if a mutex error is detected.
(issue ASTERISK-19463)
Reported by: mjordan
Tesetd by: mjordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@376586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Both hashtest and hashtest2 are manual testing apps that thrash hash
tables (hashtab and ao2 containers, respectively), by spinning up
several threads that randomly insert, delete, lookup and iterate over
the hash table. If the app doesn't crash, the hash table probably passes
the test. Those utils are not a part of the typical Asterisk build, so
they do not usually get compiled. This all makes them less that useful.
This patch removes those manual test programs and replaces them with
Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It also
attempts to make the tests more deterministic.
* Rather than spinning up some number of threads that operate on the
hash table randomly, spin up four threads that concurrenly add,
remove, lookup and iterate over the hash table.
* Each thread checks the state of the hash table both during and after
execution, and indicates a test failure if things are not as expected.
* Each thread times out after 60 seconds to prevent deadlocking the unit
test run.
(closes issue ASTERISK-20505)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2189/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@376306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Makes malloc() behave like calloc(). It will return a memory block
filled with 0x55. A nonzero value.
* Makes free() fill the released memory block and boundary fence's with
0xdeaddead. Any pointer use after free is going to have a pointer
pointing to 0xdeaddead. The 0xdeaddead pointer is usually an invalid
memory address so a crash is expected.
* Puts the freed memory block into a circular array so it is not reused
immediately.
* When the circular array rotates out a memory block to the heap it checks
that the memory has not been altered from 0xdeaddead.
* Made the astmm_log message wording better.
* Made crash if the DO_CRASH menuselect option is enabled and something is
found.
* Fixed a potential alignment issue on 64 bit systems.
struct ast_region.data[] should now be aligned correctly for all
platforms.
* Extracted region_check_fences() from __ast_free_region() and
handle_memory_show().
* Updated handle_memory_show() CLI usage help.
Review: https://reviewboard.asterisk.org/r/2182/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@376029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this change, a common method for determining if a timeout
was reached was to call a function such as ast_waitfor_n() and inspect
the out parameter that told how many milliseconds were left, then use
that as the input to ast_waitfor_n() on the next go-around.
The problem with this is that in some cases, submillisecond timeouts
can occur, resulting in the out parameter not decreasing any. When this
happens thousands of times, the result is that the timeout takes much
longer than intended to be reached. As an example, I had a situation where
a 3 second timeout took multiple days to finally end since most wakeups
from ast_waitfor_n() were under a millisecond.
This patch seeks to fix this pattern throughout the code. Now we log the
time when an operation began and find the difference in wall clock time
between now and when the event started. This means that sub-millisecond timeouts
now cannot play havoc when trying to determine if something has timed out.
Part of this fix also includes changing the function ast_waitfor() so that it
is possible for it to return less than zero when a negative timeout is given
to it. This makes it actually possible to detect errors in ast_waitfor() when
there is no timeout.
(closes issue ASTERISK-20414)
reported by David M. Lee
Review: https://reviewboard.asterisk.org/r/2135/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a bridge is broken by an AMI Redirect action or the ChannelRedirect
application, an in progress DTMF digit could be stuck sending forever.
* Made simulate a DTMF end event when a bridge is broken and a DTMF digit
was in progress.
(closes issue ASTERISK-20492)
Reported by: Jeremiah Gowdy
Patches:
bridge_end_dtmf-v3.patch.txt (license #6358) patch uploaded by Jeremiah Gowdy
Modified to jira_asterisk_20492_v1.8.patch
jira_asterisk_20492_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2169/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently, if an acknowledgement of a timer fails Asterisk will not realize
that a serious error occurred and will continue attempting to use the timer's
file descriptor. This can lead to situations where errors stream to the
CLI/log file. This consumes significant resources, masks the actual problem
that occurred (whatever caused the timer to fail in the first place), and
can leave channels in odd states.
This patch propagates the errors in the timing resource modules up through
the timer core, and makes users of these timers handle acknowledgement
failures. It also adds some defensive coding around the use of timers
to prevent using bad file descriptors in off nominal code paths.
Note that the patch created by the issue reporter was modified slightly for
this commit and backported to 1.8, as it was originally written for
Asterisk 10.
(issue ASTERISK-20032)
Reported by: Jeremiah Gowdy
patches:
jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Replace links to missing text files removed in the 1.6.x series with links to the wiki. Doxygen can handle URLs fine, don't atempt to quote them. Also update the wiki link in the Readme to get everyone on the same page.
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Passing an ast_str pointer by value that then calls
ast_str_set(), ast_str_set_va(), ast_str_append(), or
ast_str_append_va() can result in the pointer originally
passed by value being invalidated if the ast_str had
to be reallocated.
This fixes places in the code that do this. Only the
example in ccss.c could result in pointer invalidation
though since the other cases use a stack-allocated ast_str
and cannot be reallocated.
I've also updated the doxygen in strings.h to include
notes about potential misuse of the functions mentioned
previously.
Review: https://reviewboard.asterisk.org/r/2161
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375025 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is used to solve an issue where a poll on a file
descriptor does not necessarily correspond to the readiness
of a FILE handle to be read.
This change makes it so that for TCP connections, we do a
recv() on the file descriptor instead.
Because TCP does not guarantee that an entire message or even
just one single message will arrive during a read, a loop has
been introduced to ensure that we only attempt to handle a
single message at a time. The tcptls_session_instance structure
has also had an overflow buffer added to it so that if more
than one TCP message arrives in one go, there is a place to
throw the excess.
Huge thanks goes out to Walter Doekes for doing extensive review
on this change and finding edge cases where code could fail.
(closes issue ASTERISK-20212)
reported by Phil Ciccone
Review: https://reviewboard.asterisk.org/r/2123
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@374905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Greenlight in #asterisk brought up that he was receiving an error message "Could
not create persistent member string, out of space" when running app_queue in
Asterisk 10. dump_queue_members() made an assumption that 8K would be enough to
store the generated string, but with queues that have large member lists this is
not always the case. This patch removes the limitation and uses ast_str instead
of a fixed sized buffer.
The complicating factor comes from the fact that ast_db_get requires a buffer
and buffer size argument, which doesn't let us pull back more than what we pass
in, so I introduced a new ast_db_get_allocated() which returns an ast_strdup()'d
copy of the value from astdb.
As an aside, I did some testing on the maximum size of data that we can store in
the BDB library we distribute and was able to store a 10MB string and retrieve
it with no problems, so I feel this is a safe patch.
Review: https://reviewboard.asterisk.org/r/2136/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@374108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_waitfordigit_full would simply pass its timeout to ast_waitfor_nandfds,
expecting it to decrement the timeout by however many milliseconds were
waited. This is a problem if it consistently waits less than 1ms. The timeout
will never be decremented, and we wait... FOREVER!
This patch makes ast_waitfordigit_full manage the timeout itself. It maintains
the previously undocumented behavior that negative timeouts wait forever.
(closes issue ASTERISK-20375)
Reported by: Mark Michelson
Tested by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/2109/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When REF_DEBUG is enabled in certain files - most notably ccss.c - the 'tag'
parameter passed to __ao2_ref_debug will be a const char *. The function
currently expects that parameter to not be const. This causes a warning
when compiling, as the const qualifier is being discarded. With dev-mode
enabled, this prevents compiling Asterisk.
This patch makes __ao2_ref_debug's tag and file parameters const.
(closes issue ASTERISK-20408)
Reported by: mjordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While working with ast_parse_arg() to perform a validity check, a segfault
occurred. The segfault occurred due to passing a NULL pointer to
ast_sockaddr_parse() from ast_parse_arg(). According to the documentation in
config.h, "result pointer to the result. NULL is valid here, and can be used to
perform only the validity checks."
This patch fixes the segfault by checking for a NULL pointer. This patch also
adds documentation to netsock2.h about why it is necessary to check for a NULL
pointer.
(Closes issue ASTERISK-20006)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
asterisk-20006-netsock-null-ptr.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/1990/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 ASTERISK-19801)
Reported by: Alec Davis
(closes issue AST-891)
Reported by: Guenther Kelleter
Tested by: Guenther Kelleter
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes to situations that could cause the CEL LINKEDID_END event to
be missed.
1) During a core stop gracefully, modules are unloaded when ast_active_channels
== 0. The LINKDEDID_END event fires during the channel destructor. This means
that occasionally, the cel_* module will be unloaded before the channel is
destroyed. It seemed generally useful to wait until the refcount of all
channels == 0 before unloading, so I added a channel counter and used it in the
shutdown code.
2) During a masquerade, ast_channel_change_linkedid is called. It calls
ast_cel_check_retire_linkedid which unrefs the linkedid in the linkedids
container in cel.c. It didn't ref the new linkedid. Now it does.
Review: https://reviewboard.asterisk.org/r/1900/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
SSL_CTX structures were allocated but never freed. This was a bigger
issue for clients than servers since new SSL_CTX structures could be
allocated for each connection. Servers, on the other hand, typically
set up a single SSL_CTX for their lifetime.
This is solved in two ways:
1. In __ssl_setup(), if a tcptls_cfg has an ssl_ctx on it, it is
freed so that a new one can take its place.
2. A companion to ast_ssl_setup() called ast_ssl_teardown() has
been added so that servers can properly free their SSL_CTXs.
(issue ASTERISK-19278)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@367002 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The method ast_tvdiff_ms attempts to calculate the difference, in milliseconds,
between two timeval structs, and return the difference in a 64-bit integer.
Unfortunately, it assumes that the long tv_sec/tv_usec members in the timeval
struct are large enough to hold the calculated values before it returns. On
64-bit machines, this might be the case, as a long may be 64-bits. On 32-bit
machines, however, a long may be less (32-bits), in which case, the calculation
can overflow.
This overflow caused significant problems in MixMonitor, which uses the method
to determine if an audio factory, which has not presented audio to an audiohook,
is merely late in providing said audio or will never provide audio. In an
overflow situation, the audiohook would incorrectly determine that an audio
factory that will never provide audio is merely late instead. This led to
situations where a MixMonitor never recorded any audio. Note that this happened
most frequently when that MixMonitor was started by the ConfBridge application
itself, or when the MixMonitor was attached to a Local channel.
(issue ASTERISK-19497)
Reported by: Ben Klang
Tested by: Ben Klang
Patches:
32-bit-time-overflow-10-2012-04-26.diff (license #6283) by mjordan
(closes issue ASTERISK-19727)
Reported by: Mark Murawski
Tested by: Michael L. Young
Patches:
32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan)
(closes issue ASTERISK-19471)
Reported by: feyfre
Tested by: feyfre
(issue ASTERISK-19426)
Reported by: Johan Wilfer
Review: https://reviewboard.asterisk.org/r/1889/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364277 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fix AMI module reload deadlock regression from ASTERISK-18479 when it
tried to fix the race between calling an AMI action callback and
unregistering that action. Refixes ASTERISK-13784 broken by
ASTERISK-17785 change.
Locking the ao2 object guaranteed that there were no active callbacks that
mattered when ast_manager_unregister() was called. Unfortunately, this
causes the deadlock situation. The patch stops locking the ao2 object to
allow multiple threads to invoke the callback re-entrantly. There is no
way to guarantee a module unload will not crash because of an active
callback. The code attempts to minimize the chance with the registered
flag and the maximum 5 second delay before ast_manager_unregister()
returns.
The trunk version of the patch changes the API to fix the race condition
correctly to prevent the module code from unloading from memory while an
action callback is active.
* Don't hold the lock while calling the AMI action callback.
(closes issue ASTERISK-19487)
Reported by: Philippe Lindheimer
Review: https://reviewboard.asterisk.org/r/1818/
Review: https://reviewboard.asterisk.org/r/1820/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch ensures that the struct defined by AST_DECLARE_APP_ARGS() is always
fully initialized. I'm not sure if this fixes any real bugs, but it silences
a bunch of warnings from coverity, and is generally a good thing to do anyway.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Calling ast_indicate()/ast_indicate_data() with the channel lock held can
result in a deadlock with a local channel because of how local channels
need to avoid deadlock.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch updates the NUMLOGLEVELS define in logger.h to 32, to match the fact
that logger.c implements 32 log levels (because of the custom log level stuff).
asterisk.c uses this define to size an array of levels per remote console.
This array is modified in ast_console_toggle_loglevel(), which is called by the
"logger set level" CLI command. While the documentation for the CLI command
doesn't make it terribly obvious, you can use this CLI command to toggle a
custom log level on a remote console, as well. However, doing so led to an
invalid array index in asterisk.c.
This array is read from any time a log message is written to a console. So,
all custom log level messages resulted in a bogus read if a remote console
was connected.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch drastically simplifies the device state aggegation code.
The old method was not only overly complex, but also made it impossible
to return AST_DEVICE_INVALID from the aggregation code. The unit test
update is as a result of fixing that bug.
The SIP change stems from a bug introduced by removing a DNS lookup
for hostname-based SIP channels.
(closes issue ASTERISK-16702)
Review: https://reviewboard.asterisk.org/r/1808/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change fixes case-sensitivity for device-specific subscriptions such that
the technology identifier is case-insensitive while the remainder of the device
string is still case-sensitive. This should also preserve the original case of
the device string as passed in to the event system. CCSS is the only feature
affected as it is the only consumer of device-specific event subscriptions.
The second part of this patch addresses similar case-sensitivity issues within
CCSS itself that prevented it from functioning correctly after the fix to the
events system.
This adds a unit test to verify that the event system works as expected.
(closes issue ASTERISK-19422)
Review: https://reviewboard.asterisk.org/r/1780/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The SIP TCP/TLS worker threads were created joinable but noone could join
them if they died on their own.
* Fix the SIP TCP/TLS worker threads to not be created joinable.
* _sip_tcp_helper_thread() only needs one parameter since the pvt
parameter is only passed in as NULL and never used.
(closes issue ASTERISK-19203)
Reported by: Steve Davies
Review: https://reviewboard.asterisk.org/r/1714/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently, when using res_srtp, once the SRTP policy has been added to the
current session the policy is locked into place. Any attempt to replace an
existing policy, which would be needed if the remote endpoint negotiated a new
cryptographic key, is instead rejected in res_srtp. This happens in particular
in transfer scenarios, where the endpoint that Asterisk is communicating with
changes but uses the same RTP session.
This patch modifies res_srtp to allow remote and local policies to be reloaded
in the underlying SRTP library. From the perspective of users of the SRTP API,
the only change is that the adding of remote and local policies are now added
in a single method call, whereas they previously were added separately. This
was changed to account for the differences in handling remote and local
policies in libsrtp.
Review: https://reviewboard.asterisk.org/r/1741/
(closes issue ASTERISK-19253)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Patches:
srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283)
(with some small modifications for this check-in)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If the res_calendar module was followed immediately by one of the
calendar tech modules and "core stop gracefully" was run, Asterisk
would crash.
This patch adds use count tracking for res_calendar so that it is
unloaded after the tech modules when shutting down gracefully. It
is now not possible to unload all the of the calendar modules via
"module unload res_calednar.so", but it is still possible to unload
them all via "module unload -h res_calendar.so".
Review: https://reviewboard.asterisk.org/r/1752/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The principle difference between libvorbisfile v1.1.2 and newer (at least
v1.2.0) is the addition of the predefined callbacks OV_CALLBACKS_xxx in
vorbis/vorbisfile.h used for ov_open_callbacks().
* Updated the configure script to detect if libvorbisfile.h declares
OV_CALLBACKS_NOCLOSE.
* Copied the declaration of OV_CALLBACKS_NOCLOSE from v1.2.0 to allow
v1.1.2 to compile.
(closes issue ASTERISK-19370)
Reported by: Jonn Taylor
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Ogg/vorbis was fairly useless as a voicemail file format because it did
not implement the seek and tell format callbacks among other problems.
Since we were already using the libvorbis and libvorbisenc libraries we
can use libvorbisfile as it is also part of the vorbis library package.
* Made use the libvorbisfile to handle the ogg/vorbis file stream. The
format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile.
(closes issue ASTERISK-16926)
Reported by: sque
Patches:
ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded by sque
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Removes references to tlsbindport from http.conf.sample and manager.conf.sample
* Properly bind to port specified in tlsbindaddr, using the default port if specified.
* On a reload, properly close socket if the service has been disabled.
A note has been added to UPGRADE.txt to indicate how ports must be set for TLS.
(closes issue ASTERISK-16959)
reported by Olaf Holthausen
(closes issue ASTERISK-19201)
reported by Chris Mylonas
(closes issue ASTERISK-19204)
reported by Chris Mylonas
Review: https://reviewboard.asterisk.org/r/1709
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
AST_AUDIOHOOK_TRIGGER_WRITE and AST_AUDIOHOOK_WANTS_DTMF were overlapping which
may have caused unintended side effects. This patch moves
AST_AUDIOHOOK_TRIGGER_WRITE, and updates AST_AUDIOHOOK_TRIGGER_MODE to reflect
the original intention.
This will affect existing modules that use these flags, so be sure to recompile
as necessary.
(closes issue ASTERISK-19246)
Reported by: feyfre
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix memory leak of vars in error paths for action_originate().
* Moved struct fast_originate_helper tech and data members to stringfields.
* Simplified ActionID header handling for fast_originate().
* Added doxygen note to ast_request() and ast_call() and the associated
channel callbacks that the data/addr parameters should be treated as const
char *.
Review: https://reviewboard.asterisk.org/r/1690/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk's dnsmgr currently takes a pointer to an ast_sockaddr and updates it
anytime an address resolves to something different. There are a couple of
issues with this. First, the ast_sockaddr is usually the address of an
ast_sockaddr inside a refcounted struct and we never bump the refcount of those
structs when using dnsmgr. This makes it possible that a refresh could happen
after the destructor for that object is called (despite ast_dnsmgr_release
being called in that destructor). Second, the module using dnsmgr cannot be
aware of an address changing without polling for it in the code. If an action
needs to be taken on address update (like re-linking a SIP peer in the
peers_by_ip table), then polling for this change negates many of the benefits
of having dnsmgr in the first place.
This patch adds a function to the dnsmgr API that calls an update callback
instead of blindly updating the address itself. It also moves calls to
ast_dnsmgr_release outside of the destructor functions and into cleanup
functions that are called when we no longer need the objects and increments the
refcount of the objects using dnsmgr since those objects are stored on the
ast_dnsmgr_entry struct. A helper function for returning the proper default SIP
port (non-tls vs tls) is also added and used.
This patch also incorporates changes from a patch posted by Timo Teräs to
ASTERISK-19106 for related dnsmgr issues.
(closes issue ASTERISK-19106)
Review: https://reviewboard.asterisk.org/r/1691/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch prevents the domain string from getting mangled during the initreqprep
step by moving the initialization to before its immediate use. It also documents
this pitfall for the ast_sockaddr_stringify functions.
(issue ASTERISK-19057)
Reported by: Yuri
Review: https://reviewboard.asterisk.org/r/1678/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351559 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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349194 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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348940 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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
