Fixes ast_rwlock_timed[rd|wr]lock for BSD and variants.
The original implementations simply wrap pthread functions, which take
absolute time as an argument. The spinlock version for systems without
those functions treated the argument as a delta. This patch fixes the
spinlock version to be consistent with the pthread version.
When compiling with BETTER_BACKTRACES enabled, Asterisk will sometimes
crash when "core show locks" is run. This happens regularly in the
testsuite since several tests run "core show locks" to help with
debugging. This seems to be a fault with libraries on certain operating
systems (notably CentOS 6.2/6.3) running on virtual machines and
utilizing gcc 4.4.6.
(closes issue ASTERISK-20090)
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Merged revisions 371690 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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In some cases, recovering lost packets using the secondary packet
recovery mechanism with UDPTL/T.38 can result in the recovery of
zero-length packets. These must be ignored or the frame generated from
them can cause segfaults and allocation failures.
(closes issue ASTERISK-19762)
(closes issue ASTERISK-19373) Reported-by: Benjamin (bulkorok) Reported-by: Rob Gagnon (rgagnon)
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Merged revisions 371544 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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When formatting documentation fields, the XML documentation parser calls
xmldoc_get_formatted. This function allocates a string buffer at the
beginning of its routine. Unfortunately, on certain code paths, it also
calls xmldoc_string_cleanup, which assumes that it will create the string
buffer. The previously allocated string buffer is then leaked by the
xmldoc_string_cleanup routine.
Now: we don't do that.
(closes issue AST-932)
Reported by: Alexander Homig
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Merged revisions 371469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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When Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now
generate TestEvent AMI events on subsystem reloads such as cdr, dnsmgr,
extconfig, etc.
(issue PQ-1126)
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Merged revisions 371436 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Add module reload instrumentation for TEST_FRAMEWORK
This adds AMI events for module reloads when Asterisk is built with
TEST_FRAMEWORK enabled and corrects generation of the module load AMI
event.
(issue PQ-1126)
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Merged revisions 371393 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r371398 | twilson | 2012-08-16 17:50:12 -0500 (Thu, 16 Aug 2012) | 13 lines
Handle integer over/under-flow in ast_parse_args
The strtol family of functions will return *_MIN/*_MAX on overflow. To
detect when an overflow has happened, errno must be set to 0 before
calling the function, then checked afterward.
(closes issue ASTERISK-20120)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2073/
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Merged revisions 371392 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK
Under certain conditions, a SIP transaction involving directmedia wouldn't
trigger a re-invite because the SDP answer was included in an ACK instead
of in a message that we would have triggered the invite with. This patch
just queues a source change control frame if the dialog is using
directmedia when we find sdp for an ACK.
(closes issue AST-913)
Reported by: Thomas Arimont
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Merged revisions 371337 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Fix bug where final queue member would not be removed from memory.
If a static queue had realtime members, then there could be a potential
for those realtime members not to be properly deleted from memory.
If the queue's members were loaded from realtime and then all the
members were deleted from the backend, then the queue would still
think these members existed. The reason was that there was a short-
circuit in code such that if there were no members found in the
backend, then the queue would not be updated to reflect this.
Note that this only affected static queues with realtime members.
Realtime queues with realtime members were unaffected by this issue.
(closes issue ASTERISK-19793)
reported by Marcus Haas
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Merged revisions 371306 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Avoid unconditional NULLing of mwipvt on relatedpeer on SIP dialog destruction
The other instance of this bug was fixed by jcolp/file in r121496. If
we are destroying a dialog only set the MWI dialog pointer on the
related peer to NULL if it is the dialog currently being destroyed.
The helper function, get_address_family_filter, in chan_sip for dns resolution
by address family was not recognizing the websockets transport and resulting in
a null pointer being sent to functions in netsock2, in an attempt to determine
if we are bound to ANY address ([::]) or not.
This patch fixes this issue by handling the transport types SIP_TRANSPORT_WS and
SIP_TRANSPORT_WSS which results in a sock address being set properly for use in
determining the address family.
(closes issue ASTERISK-20221)
Reported by: Sven Beisiegel
Tested by: Sven Beisiegel, James Mortensen
Patches:
asterisk-20221-ws-family-filter.diff uploaded by Michael L. Young (license 5026)
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r371251 | elguero | 2012-08-14 20:43:23 -0500 (Tue, 14 Aug 2012) | 4 lines
Reverting this change that was meant for branch 11.
Fix problem where incorrect pointer was checked for nullity.
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Merged revisions 371198 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r371203 | kmoore | 2012-08-13 15:04:15 -0500 (Mon, 13 Aug 2012) | 13 lines
Add test instrumentation
This adds test instrumentation for loading and unloading of modules
and for certain actions in MeetMe to be used in the testsuite or any
other consumer of AMI events. These will only be generated when
Asterisk is built with TEST_FRAMEWORK enabled.
(issue PQ-1131)
(issue PQ-1133)
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Merged revisions 371201 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Fix a couple of documentation problems in app_queue.c
* The RemoveQueueMember app made mention of options that could
be passed in, but no options are supported. I have removed the
listing of options from the documentation.
* The RQMSTATUS variable did not list "NOTDYNAMIC" as a possible
value that could be set.
(closes issue AST-949)
reported by Steve Pitts
(closes issue AST-954)
reported by Steve Pitts
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Merged revisions 371141 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Automerge script [Fri, 10 Aug 2012 17:23:50 +0000 (17:23 +0000)]
Merged revisions 371090 via svnmerge from
file:///srv/subversion/repos/asterisk/branches/10
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r371090 | may | 2012-08-10 11:46:38 -0500 (Fri, 10 Aug 2012) | 12 lines
remove ALREADYGONE flag on ooh323 call data by ooh323_indicate
(CONGESTION/BUSY) due to call hasn't gone there really.
This indication arrive from asterisk core not h.323 stack
The documentation for the x flag for MeetMe incorrectly described its
function as closing down the conference when the last marked user left.
It actually causes the users with that flag to leave the conference
when the last marked user exits. The functionality of this flag is not
changing.
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Merged revisions 370985 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r370989 | may | 2012-08-09 13:05:34 -0500 (Thu, 09 Aug 2012) | 5 lines
change opening h323 logfile with append mode instead of overwrite
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Merged revisions 370988 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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When a channel hangs up while being spied upon and the option to exit the
ChanSpy application when the spied on channel hangs up is set,
ast_autochan_destroy is not being called and therefore a reference to the spied
upon channel is not removed.
The symptom being reported was that when using func_group in the dialplan and
calling "group show channels" at the cli, the spied upon channel was still
being shown while "core show channels" showed that the channel was not up.
This patch calls ast_autochan_destroy when a spied upon channel hangs up and
the option to exit the ChanSpy application is set, removing the reference to
the channel allowing the count for the group that the spied channel was part of
to be decremented.
(closes issue ASTERISK-17515)
Reported by: Arkadiusz Malka
Tested by: Alexandr Gordeev, Michael L. Young
Patches:
asterisk-17515-destroy-autochan.diff
uploaded by Michael L. Young (license 5026)
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Merged revisions 370952 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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AST_CAUSE_NOTDEFINED is a placeholder for usage when there is no cause
information. As such, it should not be defined and translatable as a
cause.
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Merged revisions 370923 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Fix error in the "IPorHost" section of a SIP dialstring.
This is based on the review request posted by Walter Doekes
(referenced lower in the commit message)
The main fix here is to treat the IPorHost portion of the dial
string as a temporary outbound proxy. This ensures requests
get sent to the proper location.
Due to the age of the request, some parts were no longer relevant.
For instance, the request moved outbound proxy parsing code into
a single method. This is done in a previous commit, so it was not
necessary to do again.
Also, the review request fixed some errors with regards to request
routing for CANCEL and ACK requests. This has also been fixed in
more recent commits.
(closes issue ASTERISK-19677)
reported by Walter Doekes
Schedule pokes of registered SIP peers within a given timespan after SIP reload
With a large number of SIP peers registered, performing a SIP reload causes a
flood of SIP OPTIONS request packets. These are immediately sent out, and, as
responses come back, can cause peers to be flagged as 'lagged' due to handling
of the many response messages.
This fix prevents this "packet storm" and schedules the pokes for a random
time. That time varies between 1 ms and the peer's qualify time, or, if
the qualify time is unknown, the global qualifyfreq setting.
The committed patch has some very small modifications to the patch schmidts
wrote for the review.
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().
When Asterisk servers are set up back-to-back, and
direct media is to be used betweeen endpoints, it is
fairly common for the two Asterisk servers to send
direct media reinvites to each other simultaneously.
This results in 491s and ACKs being exchanged between
the servers. While the media eventually gets set up
properly, the problem is that there can be a noticeable
delay for the streams to stabilize.
This patch adds a new directmedia option called "outgoing".
With this set, an immediate direct media reinvite will only
be sent if the call direction is outgoing. For incoming
dialogs, an immediate direct media reinvite will not be sent,
but further "reactionary" direct media reinvites may be sent.
For those who are having some deja vu, that's because this
patch was originally committed to trunk since there is a
new configuration option added. After seeing a bug report
about audio being slow to set up on SIP calls, it became
apparent that this patch would be the best solution for
resolving the issue. The patch is unintrusive and will
have no effect unless the option is explicitly enabled.
(closes issue AST-896)
reported by Thomas Arimont
(closes issue ASTERISK-19857)
reported by Matt Jordan
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Merged revisions 370618 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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res_agi: Add message indicating need for \n character in verbose message
The while loop responsible for reading AGI messages from a fastAGI service
can end up looping indefinitely when an AGI script fails to indicate the end
of a message with a \n character. This patch adds an indication that we are
expecting a \n character to end the message to make it more clear to users
that this is necessary if they are receiving this warning over and over.
Terry Wilson [Wed, 25 Jul 2012 03:45:20 +0000 (03:45 +0000)]
Revert a change that broke compilation
1) There is no such function as ast_ref()
2) The patch was originally credited as the one uploaded by Guenther
Kelleter (license 6372) via issue AST-921, but the patch committed
was not the patch referenced on the issue.
3) Guenther Kelleter's patch was actually correct. It moved the
ast_free above the presencechange_cleanup label. I am not
committing his change as it is not technically necesary--calling
ast_free(NULL) is perfectly safe and I worry that moving the
ast_free outside of the label could lead to future bugs if
someone ever adds another failure conditional and expects
'goto presencechange_cleanup;' to clean up after everything.
Revision 370205 added the use of a datastore attached to a dummy channel to
resolve a memory leak, but ast_dummy_channel_destructor() in this branch did
not free datastores, resulting in a continued (but slightly smaller) memory
leak. This patch backports the change to free said datastores from the Asterisk
trunk.
(related to issue AST-916)
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Merged revisions 370360 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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The current implementation of RFC 2833 DTMF handling in res_rtp_asterisk will,
if a packet arrives out of order, drop the packet. This is to prevent
duplicate ton generation in the Asterisk core. Since the RTP layer does not
buffer data itself, this is the only option the RTP layer currently has for
handling packets that arrive out of order.
For the most part, this doesn't matter. For a particular digit, so long as a
BEGIN packet arrives before the first END packet, the digit will be produced.
If subsequent BEGIN packets arrive interleaved with the ENDs, they will be
dropped; likewise, if the BEGIN or END packets themselves are out of order,
those packets are dropped but sufficient information is conveyed to the
Asterisk core to produce the appropriate digit.
For certain sequences of DTMF packets - most notably when, for a particular
digit, an END packet arrives before any BEGIN packet for that digit - this
is a real problem. When an END arrives before any BEGINs, the END packet is
dropped - but at the same time, it causes subsequent BEGIN packets for that
digit to be ignored. When the next in order END packet arrives, it too is
dropped - Asterisk believes that there was no initial BEGIN.
The solution this patch provides is to trust the END packet to convey the
information needed for the Asterisk core to produce the DTMF digit. If we
receive an END packet, and it:
* Has a timestamp greater then the last timestamp received from an END
packet
* Does not have the same sequence number as the last received sequence
number (and is thus not an END packet retransmission)
Then we send the END frame up to the Asterisk core. It contains enough
DTMF information for Asterisk to produce the digit.
On the other hand, if we receive a BEGIN or continuation packet that occurs
with a timestamp equal to or less then the last END timestamp, then we've
received something out of order - but we already have received enough
information to produce the digit. These packets are dropped.
Much thanks goes to Olle Johansson (oej) for providing the idea for this
solution.
Review: https://reviewboard.asterisk.org/r/2033/
(issue ASTERISK-18404)
Reporter: Stephane Chazelas
Tested by: Matt Jordan
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Fix compilation error when MALLOC_DEBUG is enabled
To fix a memory leak in CEL, a channel datastore was introduced whose
destruction function pointer was pointed to the ast_free macro. Without
MALLOC_DEBUG enabled this compiles as fine, as ast_free is defined as free.
With MALLOC_DEBUG enabled, however, ast_free takes on a definition from a
different place then utils.h, and became undefined. This patch resolves this
by using a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this
calls ast_free; when MALLOC_DEBUG is not enabled, this is defined to be
ast_free, which is defined to be free.
(issue AST-916)
Reported by: Thomas Arimont
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Resolve severe memory leak in CEL logging modules.
A customer reported a significant memory leak using Asterisk 1.8. They
have tracked it down to ast_cel_fabricate_channel_from_event() in
main/cel.c, which is called by both in-tree CEL logging modules
(cel_custom.c and cel_sqlite3_custom.c) for each and every CEL event
that they log.
The cause was an incorrect assumption about how data attached to an
ast_channel would be handled when the channel is destroyed; the data
is now stored in a datastore attached to the channel, which is
destroyed along with the channel at the proper time.
Ensure that all ast_datastore_info structures are 'const'.
While addressing a bug, I came across a instance of 'struct ast_datastore_info'
that was not declared 'const'. Since the API already expects them to be
'const', this patch changes the declarations of all existing instances
that were not already declared that way.
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Merged revisions 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Prevent double uri_escaping in chan_sip when pedantic is enabled
If pedantic mode is enabled, outbound invites will have double-escaped
contacts. This avoids setting an already-escaped string into a field
where it is expected to be unescaped.
(closes issue ASTERISK-20023) Reported-by: Walter Doekes
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Merged revisions 369993 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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The documentation for DEC in func_math.c was incorrect. Looks like a copy and
paste error.
(Closes issue ASTERISK-20095)
Reported by: Billy Chia
Tested by: Michael L. Young
Patches:
func_math.patch uploaded by Billy Chia (license 6381)
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Merged revisions 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Allow the REALTIME() function to report errors back to the caller.
Also, do more error checking on the arguments specified to the REALTIME()
function and clarify the documentation. While I was editing the file, a
few coding guidelines fixups, as well.
chan_sip: Fix small behavioral change accidentally introduced in r369750
When removing the warning for AST_CONTROL_FLASH from sip_indicate, I also
inadvertently changed the return value, which would likely make the indication
not be sent in audio. This fixes that while still removing the warning message.
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Merged revisions 369792 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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chan_sip: Add case for FLASH control frames so that we don't display a warning.
chan_sip channels can receive flash control frames when connected to analog
phones and possibly for other reasons. There really isn't a reason to warn when
these frames are received, we can safely ignore them.
Patches:
dahdi_sip_flash.diff uploaded by Jonathan Rose (license 6182)
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Merged revisions 369750 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Remove a superfluous and dangerous freeing of an SSL_CTX.
The problem here is that multiple server sessions share
a SSL_CTX. When one session ended, the SSL_CTX would be
freed and set NULL, leaving the other sessions unable to
function.
The code being removed is superfluous because the SSL_CTX
structures for servers will be properly freed when ast_ssl_teardown
is called.
(closes issue ASTERISK-20074)
Reported by Trevor Helmsley
Patches:
ASTERISK-20074.diff uploaded by Mark Michelson (license #5049)
Testers:
Trevor Helmsley
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Merged revisions 369731 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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The bridge thread was exiting but was never being
reaped using pthread_join(). This has been fixed now
by calling pthread_join() in ast_bridge_destroy().
(closes issue ASTERISK-19834)
Reported by Marcus Hunger
The heard and deleted arrays in the voicemail state structure were not
handled properly following the memory leak fix in r354890 and a fix for
an invalid free in r356797. This could result in accessing and writing
into freed memory. The allocation for these arrays has been reworked
to avoid the possibility of invalid frees, access of freed memory, and
crashes that were occurring as a result of this.
Locking around accesses and modifications of the voicemail state
structure members dh_arraysize, heard, and deleted has been added to
prevent simultaneous modification and access when IMAP storage is in
use. If IMAP storage is not in use, this locking is not compiled in.
Do not send a BYE when a provisional response arrives during a re-INVITE
Commits r369557 and r369579 were done to improve handling of re-INVITEs
when the UA that was supposed to receive the re-INVITE fails to respond.
A limitation of those patches occurred when a UA sent a provisional
response to the re-INVITE. This triggered a sending of a BYE in
check_pending. This patch tweaks the handling of the re-INVITE such that
a BYE is not sent in response to those messages.
(issue ASTERISK-19992)
Reported by: Steve Davies
Tested by: Steve Davies
patches:
(reinvite_tweak.diff license #5012 by Steve Davies)
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Merged revisions 369626 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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More improvements to re-INVITEs timing out after a provisional response
There is no need to call check_pendings() on a final response to an INVITE
when destroying the scheduler entry as it will be done later during normal
processing.
(issue ASTERISK-19992)
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Merged revisions 369579 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Better handle re-INVITEs with provisional but no final repsonses
A previous attempt at fixing this issue had negative side effects related
to attended transfers which this patch should resolve. Many thanks to
Steve Davies for all of the good suggestions and testing.
(closes issue ASTERISK-19992)
Reported by: Steve Davies
Tested by: Steve Davies, Terry Wilson
Review: https://reviewboard.asterisk.org/r/2009/
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Merged revisions 369557 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Clean up after a reinvite that never gets a final response
The basic problem is that if a re-INVITE is sent by Asterisk and it receives a
provisional response, but no final response, then the dialog is never torn
down. In addition to leaking memory, this also leaks file descriptors and will
eventually lead to Asterisk no longer being able to process calls.
This patch just keeps track of whether there is an outstanding re-INVITE, and if
there is goes ahead and cleans up everything as though there was no outstanding
reinvite.
(closes issue ASTERISK-19992)
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Merged revisions 369436 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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When res_adsi is unloaded, it removes the ADSI functions that it previously installed
by passing a NULL adsi_funcs pointer to ast_adsi_install_funcs. This function was not
checking whether or not the adsi_funcs pointer passed in was NULL before dereferencing
it to check whether or not the version of the functions matches what the core was
expecting it.
This patch makes it so that the version is only checked if a potentially valid adsi_funcs
pointer was passed in. Passing in NULL removes the installed functions, bypassing the
version check.
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Merged revisions 369390 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Fix incorrect duration reporting in CDRs created in batch mode
Certain places in core/cdr.c would, if the duration value were 0, calculate the
duration as being the delta between the current time and the time at which the
CDR record was started. While this does not typically cause a problem in
non-batch mode, this can cause an issue in batch mode where CDR records are
gathered and written long after those calls have ended. In particular, this
affects calls that were never answered, as those are expected to have a duration
of 0. Often, this would result in CDR logs with a significant number of calls
with lengthy durations, but dispositions of "BUSY".
Note that this does not affect cdr_csv, as that backend does not use
ast_cdr_getvar and instead directly reports the duration value. The affected
core backends include cdr_apative_odbc and cdr_custom; other extended or
deprecated CDR backends may potentially still directly manipulate the duration
values.
(issue ASTERISK-19860)
Reported by: Thomas Arimont
(issue AST-883)
Reported by: Thomas Arimont
Tested by: Matt Jordan
Re-fix how local tag is generated when sending a 481 to an INVITE.
Match our local tag to whatever to-tag was sent in the initial INVITE.
Because the size of the to-tag may not fit in the buffer in the sip_pvt,
it has been changed to a string field.
(closes issue ASTERISK-19892)
reported by Walter Doekes
Forgot to svn add this file in my last commit.
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Merged revisions 369323-369324 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r369328 | rmudgett | 2012-06-25 10:59:28 -0500 (Mon, 25 Jun 2012) | 15 lines
Fix Bridge application occasionally returning to the wrong location.
* Fix do_bridge_masquerade() getting the resume location from the zombie
channel. The code must not touch a clone channel after it has masqueraded
it. The clone channel has become a zombie and is starting to hangup.
Be more consistent with the return code for requests received from invalid domain.
When Asterisk receives an INVITE from an external domain when allowexternaldomains=no
send a 403 instead of a 404. This is consistent with Asterisk's behavior when receiving
a REGISTER in this situation.
(Closes issue ASTERISK-19601)
Reported by Matthew Jordan
Patches:
ASTERISK-19601-no401.patch uploaded by Mark Michelson (License #5049)
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Merged revisions 369302 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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A sip_pvt may not have relatedpeer set if a call doesn't match up
with a peer. If there is no relatedpeer, there is no direct media
ACL to apply, so just return that it is allowed.
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Merged revisions 369214 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Don't parse media stream state for SIP video streams
The sendonly/recvonly/sendrecv/inactive media stream attributes were
parsed for video, but nothing was ever done with them. With this code
removed, an UNSUPPORTED message is produced when these attributes are
used in conjunction with a video stream which is the better behavior
since they were never really supported in the first place.
........
Merged revisions 369195 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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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)
Fix request routing issue when outboundproxy is used.
Asterisk was incorrectly setting the destination of CANCELs
and ACKs for error responses to the URI of the initial INVITE.
This resulted in further requests, such as INVITEs with authentication
credentials, to be routed incorrectly. Instead, when these CANCEL
or ACKs are to be sent, we should simply keep the destination the
same as what it previously was. There is no need to alter it any.
(closes issue ASTERISK-20008)
Reported by Marcus Hunger
Patches:
ASTERISK-20008.patch uploaded by Mark Michelson (license #5049)
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Merged revisions 369066 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Add support-level indications to many more source files.
Since we now have tools that scan through the source tree looking for files
with specific support levels, we need to ensure that every file that is
a component of a 'core' or 'extended' module (or the main Asterisk binary)
is explicitly marked with its support level. This patch adds support-level
indications to many more source files in tree, but avoids adding them to
third-party libraries that are included in the tree and to source files
that don't end up involved in Asterisk itself.
........
r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines
Add a script to enable finding source files without support-levels defined.
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Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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AST-2012-009: Fix crash in chan_skinny due to Key Pad Button Message handling
AST-2012-008 (r367844) fixed a denial of service attack exploitable in the
Skinny channel driver that occurred when certain messages are sent after a
previously registered station sends an Off Hook message. Unresolved in that
patch is an issue in the Asterisk 10 releases, wherein, if a Station Key
Pad Button Message is processed after an Off Hook message, the channel driver
will inappropriately dereference a NULL pointer.
This patch fixes those places where the message handling or the channel
callback functions would attempt to dereference the line's pointer to the
device.
(issue ASTERISK-19905)
Reported by: Christoph Hebeisen
Tested by: mjordan, Christoph Hebeisen
Patches:
AST-2012-009-10.diff uploaded by mjordan (license 6283)
........
Revert Makefile change to remove embedding res_adsi.so
The change has resulted in a linking error for certain versions
of GCC. This is much worse than the original issue, so for now,
temporarily revert the change. A more thorough change will be
sought out.
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Merged revisions 368927 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Mark res_smdi/res_adsi as 'core' supported modules
Recently, various issues surrounding weak attributes have caused problems with
modules that rely on that feature to be enabled in menuselect. This includes
app_voicemail and chan_dahdi, as they both rely upon res_smdi and res_adsi,
which, in certain circumstances, may not be enabled by default in menuselect.
Because res_smdi/res_adsi are dependencies for chan_dahdi/app_voicemail, this
patch marks both as 'core' supported modules. This will allow both
app_voicemail and chan_dahdi to be enabled as well, regardless of whether or
not that system supports weak attributes.
(issue AST-900)
Reported by: Thomas Arimont
(issue AST-885)
Reported by: Denis Alberto Martinez
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Merged revisions 368894 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r368899 | mmichelson | 2012-06-13 16:13:30 -0500 (Wed, 13 Jun 2012) | 19 lines
Fix a deadlock that occurs when func_volume is used on a local channel.
This was discovered by trying to perform a call forward to an extension
that makes use of func_volume. When the local channel is optimized away,
the datastore on the local;2 channel would have its audiohook destroyed
rather than detaching the audiohook from the channel and then destroying
it.
With this patch, func_volume's datastore destructor takes the proper
route of detaching the audiohook and then destroying it.
(closes issue ASTERISK-19611)
reported by Volker Sauer
Patches:
ASTERISK-19611.patch uploaded by Mark Michelson (license #5049)
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Merged revisions 368898 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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If a directory already exists, performing a 'make install' will remove the
permissions associated with the current directory and replace them with the
permissions of the user executing the install.
This patch changes this behavior to only perform an install on the directory
if the directory does not exist. Thus, if a user later changes the permissions
on that directory, those permissions will be preserved in subsequent installs.
Review: https://reviewboard.asterisk.org/r/1986
Review: https://reviewboard.asterisk.org/r/1864
(closes issue ASTERISK-19492)
Reported by: Karl Fife
Tested by: Paul Belanger, Tilghman Lesher
patches:
ASTERISK-19492 by pabelanger
(uploaded by mjordan)
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Merged revisions 368830 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r368853 | mjordan | 2012-06-13 09:30:34 -0500 (Wed, 13 Jun 2012) | 11 lines
Do not install empty directories; add ASTLIBDIR
r368830 modified the installation script to only create a directory if that
directory does not exist. If some directory variable was empty, it would attempt
to create the empty location. It also failed to create the ASTLIBDIR directory.
This patch fixes it such that the correct directories are made and only created if
a value specifying them actually exists.
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Merged revisions 368852 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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(closes issue ASTERISK-19801)
Reported by: Alec Davis
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Merged revisions 368759 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r368808 | mmichelson | 2012-06-12 10:37:38 -0500 (Tue, 12 Jun 2012) | 15 lines
Set the Caller ID "tag" on peers even if remote party information is present.
On incoming calls, we were setting the cid_tag on the dialog only if there was
no remote party information (Remote-Party-ID or P-Asserted-Identity) present.
The Caller ID tag is an invented parameter, though, and should be set no matter
the circumstance.
(closes issue ASTERISK-19859)
Reported by Thomas Arimont
(closes issue AST-884)
Reported by Trey Blancher
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Merged revisions 368807 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 368721,368739,368760,368808 from http://svn.asterisk.org/svn/asterisk/branches/10
Matthew Jordan [Tue, 12 Jun 2012 13:59:28 +0000 (13:59 +0000)]
Fix deadlock in SIP transfers that involve a REFER request
In r367163, "send to voicemail" functionality was added to the SIP channel
driver. This required updating the party redirecting information for the
channel based on the headers provided in the REFER request. When the
redirecting party information is updated on the channel, a call to
ast_indicate_data occurs. Because handle_request_refer still had the sip_pvt
locked, a deadlock could occur between the pbx_thread and the do_monitor thread
servicing the REFER request.
This patch preserves the proper locking order between the channel and the
sip_pvt by ensuring that the sip_pvt is unlocked prior to updating the party
redirecting information on the channel.
(closes issue AST-903)
Reported by: Matt Jordan
patches:
jira_ast_903_trunk.patch by rmudgett (license 5621)
Fix a specific scenario where ACKs are not matched.
If a dialog-starting INVITE contains a to-tag, then Asterisk
will respond with a 481. In this case, the resulting incoming
ACK would not be matched, so Asterisk would continue retransmitting
the 481 until the transaction times out.
There were two issues. Asterisk, upon creating a sip_pvt would generate
a local tag. However, when the time came to transmit the 481, since there
was a to-tag in the INVITE, Asterisk would place this original to-tag
in the 481 response. When the ACK came in, Asterisk would attempt to
match the to-tag in the ACK to the generated local tag. Unfortunately,
Asterisk never actually transmitted a response with the generated local
tag, so the to-tag in the ACK would not match.
The other problem was that when the 481 was sent, nothing was set
on the sip_pvt to indicate what CSeq is expected in the ACK.
To fix the first problem, we zero out the to-tag seen in the incoming
INVITE. This way, Asterisk, when time to send a response, will send
its generated local tag instead.
To fix the second problem, we set the sip_pvt's pendinginvite to the
CSeq of the INVITE when we send a 481.
(closes issue ASTERISK-19892)
Reported by Mark Michelson
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Merged revisions 368625 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r368645 | rmudgett | 2012-06-06 16:32:09 -0500 (Wed, 06 Jun 2012) | 17 lines
Fix POTS flash hook to orignate a second call deadlock.
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.
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r365155 | may | 2012-05-03 09:27:00 -0500 (Thu, 03 May 2012) | 11 lines
Fix coverity static analysis warning, allocate full ie structure
instead of without data buffer
(close issue ASTERISK-19674)
Reported by: Matt Jordan
Patches:
ASTERISK-19674.patch (License #5415)
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Merged revisions 365143 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r365160 | may | 2012-05-03 10:01:14 -0500 (Thu, 03 May 2012) | 11 lines
Fix warning of Coverity Static analysis, change H225ProtocolIdentifier
from value to pointer per functions that use this.
(close issue ASTERISK-19670)
Reported by: Matt Jordan
Patches:
ASTERISK-19670.patch (License #5415)
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Merged revisions 365159 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r365299 | mmichelson | 2012-05-04 10:51:04 -0500 (Fri, 04 May 2012) | 12 lines
Fix core FINDING 2, FINDING 3, and FINDING 4 from Coverity's CONSTANT_EXPRESSION_RESULT report.
These three all are in RTP code that attempts to print the number of sequence number cycles
in an RTCP RR report. The code was masking out the upper 16 bits and then shifting the number
right by 16 bits. This led to an all zero result in all cases. The fix is to do the shift without
the bit masking.
(issue ASTERISK-19649)
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Merged revisions 365298 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r365320 | rmudgett | 2012-05-04 11:28:06 -0500 (Fri, 04 May 2012) | 30 lines
Fix local channel chains optimizing themselves out of a call.
* Made chan_local.c:check_bridge() check the return value of
ast_channel_masquerade(). In long chains of local channels, the
masquerade occasionally fails to get setup because there is another
masquerade already setup on an adjacent local channel in the chain.
* Made the outgoing local channel (the ;2 channel) flush one voice or
video frame per optimization attempt.
* Made sure that the outgoing local channel also does not have any frames
in its queue before the masquerade.
* Made do the masquerade immediately to minimize the chance that the
outgoing channel queue does not get any new frames added and thus
unconditionally flushed.
* Made block indication -1 (Stop tones) event when the local channel is
going to optimize itself out. When the call is answered, a chain of local
channels pass down a -1 indication for each bridge. This blizzard of -1
events really slows down the optimization process.
(closes issue ASTERISK-16711)
Reported by: Alec Davis
Tested by: rmudgett, Alec Davis
Review: https://reviewboard.asterisk.org/r/1894/
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Merged revisions 365313 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r365399 | kmoore | 2012-05-04 17:15:05 -0500 (Fri, 04 May 2012) | 13 lines
Fix many issues from the NULL_RETURNS Coverity report
Most of the changes here are trivial NULL checks. There are a couple
optimizations to remove the need to check for NULL and outboundproxy parsing
in chan_sip.c was rewritten to avoid use of strtok. Additionally, a bug was
found and fixed with the parsing of outboundproxy when "outboundproxy=," was
set.
(Closes issue ASTERISK-19654)
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Merged revisions 365398 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r365475 | mjordan | 2012-05-07 13:39:10 -0500 (Mon, 07 May 2012) | 20 lines
Support VoiceMail d() option when extension does not exist in channel's context
The VoiceMail d([c]) option is documented to accept digits for a new extension
in context <c>, if played during the greeting. This option works fine if the
extension being redirected to has an extension with the same initial digit in
the channel's current context. If that digit did not happen to exist in some
extension, a dialplan match would fail and the user would not be redirected.
This patch fixes it such that if the <c> option is used, the extensions are
matched in that context as opposed to the caller's original context.
Merged revisions 365474 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r365478 | rmudgett | 2012-05-07 13:43:08 -0500 (Mon, 07 May 2012) | 5 lines
Fix type punned compiler warning in test_config.c
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Merged revisions 365476 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r365575 | mmichelson | 2012-05-08 10:51:13 -0500 (Tue, 08 May 2012) | 22 lines
Send more accurate identification information in dialog-info SIP NOTIFYs.
This uses the calling channel's caller ID and connected line information
to populate the remote and local identities in the dialog-info NOTIFY when
an extension is ringing.
There is a bit of an oddity here, and that is that we seed the remote target
with the To header of the outbound call rather than the from header. This
is because it was reported that seeding with the from header caused hints
to be broken with certain SNOM devices. A comment has been added to the code
to explain this.
(closes issue ASTERISK-16735)
reported by Maciej Krajewski
patches:
local_remote_hint2.diff uploaded by Mark Michelson (license #5049)
16735_tweak1.diff uploaded by Mark Michelson (license #5049)
Tested by Niccolo Belli
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Merged revisions 365574 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r365632 | rmudgett | 2012-05-08 13:08:01 -0500 (Tue, 08 May 2012) | 13 lines
* Fix accept/decline DTMF buffer overwrite in FollowMe.
* Made use MAX_YN_STRING define to make all accept/decline DTMF buffers
the same size. Just using 20 isn't good enough when someone didn't get
the memo.
* Fix stupid use of a global variable in FollowMe. (ynlongest)
* Fix bit field declarations in FollowMe.
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Merged revisions 365631 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r365701 | rmudgett | 2012-05-08 15:25:08 -0500 (Tue, 08 May 2012) | 12 lines
* Fix FollowMe memory leak on error paths in app_exec().
* Fix FollowMe leaving recorded caller name file on error paths in
app_exec().
* Use correct buffer dimension define in struct call_followme.moh[] and
struct fm_args.namerecloc[]. This fixes unexpected namerecloc filename
length restriction.
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Merged revisions 365692 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r365898 | mmichelson | 2012-05-09 11:15:28 -0500 (Wed, 09 May 2012) | 29 lines
Prevent sip_pvt refleak when an ast_channel outlasts its corresponding sip_pvt.
chan_sip was coded under the assumption that a SIP dialog with an owner channel
will always be destroyed after the owner channel has been hung up.
However, there are situations where the SIP dialog can time out and auto destruct
before the corresponding channel has hung up. A typical example of this would be
if the 'h' extension in the dialplan takes a long time to complete. In such cases,
__sip_autodestruct() would complain about the dialog being auto destroyed with
an owner channel still in place. The problem is that even once the owner channel
was hung up, the sip_pvt would still be linked in its ao2_container because nothing
would ever unlink it.
The fix for this is that if __sip_autodestruct() is called for a sip_pvt that still
has an owner channel in place, the destruction is rescheduled for 10 seconds in the
future. This will continue until the owner channel is finally hung up.
(closes issue ASTERISK-19425)
reported by David Cunningham
Patches:
ASTERISK-19425.patch uploaded by Mark Michelson (License #5049)
(closes issue ASTERISK-19455)
reported by Dean Vesvuio
Tested by Dean Vesvuio
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Merged revisions 365896 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r365990 | jrose | 2012-05-09 14:12:32 -0500 (Wed, 09 May 2012) | 18 lines
Block on frameout if the hardware has enough samples to complete a frame.
Fixes some problems with skipping audio in elaborate scenarios involving
multiple codecs by making codec_dahdi operate in a more synchronous
fashion similar to codec_g729. This change also fixes the use of file
conversion tools from Asterisk's CLI. This change may cause the thread
responsible for transcoding audio to block briefly (Shaun Ruffell describes
this as 'several milliseconds') while waiting for the hardware transcoder.
Merged revisions 365989 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r366049 | jrose | 2012-05-10 10:43:06 -0500 (Thu, 10 May 2012) | 9 lines
Coverity Report: Fix issues for error type UNINIT in Core supported modules
(issue ASTERISK-19652)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1909/
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Merged revisions 366048 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r366053 | mmichelson | 2012-05-10 11:13:06 -0500 (Thu, 10 May 2012) | 9 lines
Close the proper tcptls_session when session creation fails.
(issue AST-998)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
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Merged revisions 366052 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r366106 | jrose | 2012-05-10 11:55:22 -0500 (Thu, 10 May 2012) | 9 lines
Coverity Report: Fix issues for error type CHECKED_RETURN for core
(issue ASTERISK-19658)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1905/
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Merged revisions 366094 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r366168 | kmoore | 2012-05-10 15:54:08 -0500 (Thu, 10 May 2012) | 13 lines
Resolve FORWARD_NULL static analysis warnings
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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Merged revisions 366167 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r366241 | rmudgett | 2012-05-10 18:42:43 -0500 (Thu, 10 May 2012) | 7 lines
* Made ast_change_name() hold the channels container lock while changing the channel name.
* Eliminate redundant list not empty check in clone_variables().
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Merged revisions 366240 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r366297 | russell | 2012-05-11 18:59:35 -0500 (Fri, 11 May 2012) | 19 lines
format_mp3: Fix a possible crash mp3_read().
This patch fixes a potential crash in mp3_read() by not assuming that
dbuf has enough data to finish filling up the output buffer. The patch
also makes sure that the dbuf state gets reset after we know we read
everything out of it already.
In passing, this patch includes some other cleanups of this module,
including stripping trailing whitespace, formatting fixes based on
coding guidelines, and removing a number of unused members from the
private state struct.
(closes issue ASTERISK-19761)
Reported by: Chris Maciejewsk
Tested by: Chris Maciejewsk
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Merged revisions 366296 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r366390 | mmichelson | 2012-05-14 14:16:36 -0500 (Mon, 14 May 2012) | 25 lines
Fix broken reinvite glare scenario.
To make a long story short, reinvite glares were broken
because Asterisk would invert the To and From headers
when ACKing a 491 response.
The reason was because the initreq of the dialog was being
changed to the incoming glared reinvite instead of being
set to the outgoing glared reinvite. This change has three
parts
* In handle_incoming, we never will reject an ACK because it
has a to-tag present, even if we think the request may be out
of dialog.
* In handle_request_invite, we do not change the initreq when
receiving a reinvite to which we will respond with a 491.
* In handle_request_invite, several superflous settings up
pendinginvite have been removed since this is dones automatically
by transmit_response_reliable
Merged revisions 366389 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r366412 | mmichelson | 2012-05-14 15:06:58 -0500 (Mon, 14 May 2012) | 19 lines
Fix two more coverity constant expression result findings.
These correspond to findings 0 and 1 in the core findings of
ASTERISK-19649.
After contacting Mark Spencer, he was unsure of what the intent
behind these lines of code were, so they are being axed.
For Asterisk 1.8 and 10, the output of debugging DUNDi frames
will not be changed, but for trunk the "Retry" portion will
be omitted since it does not properly distinguish retransmissions
from initial frames.
(closes issue ASTERISK-19649)
Reported by Matthew Jordan
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Merged revisions 366409 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r366591 | jrose | 2012-05-15 15:44:59 -0500 (Tue, 15 May 2012) | 15 lines
chan_sip: Check the right channel's host address for directmediapermit/deny
Prior to this patch, when checking the addresses for directmediapermit and
denydirectmediadeny, Asterisk would check the host address of the channel
permit/deny was specified, which defers from the expectations of both
our users and the development team. Instead, directmediapermit/deny now
checks against the address of the channel that the peer with the ACL is
connected to.
Merged revisions 366547 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r366598 | mmichelson | 2012-05-15 18:39:06 -0500 (Tue, 15 May 2012) | 8 lines
Correct misuse of ast_strip_quoted() when getting a Diversion header's reason parameter.
The use here was assuming that the pointer would be updated, but the updated string
is actually returned by ast_strip_quoted() instead.
........
Merged revisions 366597 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r366741 | mjordan | 2012-05-17 07:57:30 -0500 (Thu, 17 May 2012) | 23 lines
Fix checking bounds of array index after using it; improper sizeof
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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Merged revisions 366740 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r366792 | jrose | 2012-05-17 09:41:13 -0500 (Thu, 17 May 2012) | 10 lines
chan_sip: Fix missed locking of opposing pvt for directmedia acl from r366547
It also required deadlock avoidance since two sip_pvts structs needed to be
locked simultaneously. Trunk handles it differently, so this is a 1.8 and 10
patch only.
........
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
Merged revisions 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r366884 | kmoore | 2012-05-18 09:18:47 -0500 (Fri, 18 May 2012) | 9 lines
Reorder and renumber tests appropriately
It appears that a patch did not apply properly when adding tests 12 and
13 and test 11 was duplicated. These tests have been reordered and
renumbered such that they make sense.
........
Merged revisions 366882 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r366948 | mjordan | 2012-05-18 10:45:42 -0500 (Fri, 18 May 2012) | 20 lines
Fix more memory leaks
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
Merged revisions 366944 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r367003 | mmichelson | 2012-05-18 12:00:14 -0500 (Fri, 18 May 2012) | 19 lines
Fix memory leak of SSL_CTX structures in TLS core.
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)
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Merged revisions 367002 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r367028 | mmichelson | 2012-05-18 12:50:18 -0500 (Fri, 18 May 2012) | 18 lines
Address MISSING_BREAK static analysis reports some more.
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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Merged revisions 367027 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r367267 | twilson | 2012-05-22 11:17:46 -0500 (Tue, 22 May 2012) | 14 lines
Resolve crash in subscribing for MWI notifications
ASTOBJ_UNREF sets the variable to NULL after unreffing it, so the variable
should definitely not be used after that. To solve this in the two cases
that affect subscribing for MWI notifications, we instead save the ref
locally, and unref them in the error conditions.
(closes issue ASTERISK-19827)
Reported by: B. R
Review: https://reviewboard.asterisk.org/r/1940/
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Merged revisions 367266 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r367299 | twilson | 2012-05-22 12:21:51 -0500 (Tue, 22 May 2012) | 21 lines
Fix race condition for CEL LINKEDID_END event
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.
Merged revisions 367292 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r367369 | mjordan | 2012-05-23 08:25:04 -0500 (Wed, 23 May 2012) | 26 lines
Re-add LastMsgsSent value for SIP peers
Previously, MWI logic utilized a counter called 'lastmsgssent' to know whether
or not MWI NOTIFY requests had been sent to a specific peer. When MWI
notifications were changed to use the internal event framework, this value was
no longer needed for its original purpose. Hence, it was no longer updated
with the new/old message counts for a peer. The value was previously removed
for Asterisk 10; however, since it was still present in Asterisk 1.8 and still
useful for reporting purposes, it was decided to re-add the value.
This patch re-adds the 'LastMsgsSent' field in the response to an AMI/CLI 'sip
show peer [peer]' command, and makes it so that the value of lastmsgssent is
updated appropriately. The value should now display the new/old message counts
for a particular peer.
(closes issue ASTERISK-17866)
Reported by: Steve Davies
patches by:
ast-17866-rb1272.patch (License #5041 by irroot)
Modified slightly for this commit
The AST_CONTROL_HOLD MOH class from the WaitExten application can now be
queued onto a channel, passed over local channels with the /m option, and
passed over IAX channels.
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Merged revisions 367469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r367562 | mjordan | 2012-05-24 08:32:33 -0500 (Thu, 24 May 2012) | 24 lines
Fix crash in ConfBridge when user announcement is played for more than 2 users
A patch introduced in r354938 made it so that ConfBridge would not attempt to
play sound files if those files did not exist. Unfortunately, ConfBridge uses
the same underlying function, play_sound_helper, to playback both sound files
and numbers to callers. When a number is being played back, the name of the
sound file is expected to be NULL. This NULL value was passed into a function
that tested for the existance of a sound file and is not tolerant to NULL
file names, causing a crash.
This patch fixes the behavior, such that if a sound file does not exist we
do not attempt to play it, but we only attempt that check if the a sound file
was specified in the first place. If a sound file was not specified, we use
the 'play number' logic in the helper function.
Fix Dial I option ignored if dial forked and one fork redirects.
The Dial and Queue I option is intended to block connected line updates
and redirecting updates. However, it is a feature that when a call is
locally redirected, the I option is disabled if the redirected call runs
as a local channel so the administrator can have an opportunity to setup
new connected line information. Unfortunately, the Dial and Queue I
option is disabled for *all* forked calls if one of those calls is
redirected.
* Make the Dial and Queue I option apply to each outgoing call leg
independently. Now if one outgoing call leg is locally redirected, the
other outgoing calls are not affected.
* Made Dial not pass any redirecting updates when forking calls.
Redirecting updates do not make sense for this scenario.
* Made Queue not pass any redirecting updates when using the ringall
strategy. Redirecting updates do not make sense for this scenario.
* Fixed deadlock potential with chan_local when Dial and Queue send
redirecting updates for a local redirect.
* Converted the Queue stillgoing flag to a boolean bitfield.
Merged revisions 367678 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r367731 | elguero | 2012-05-24 21:29:26 -0500 (Thu, 24 May 2012) | 20 lines
Fix pvt_sip for inbound call to use peer's allowtransfer setting
The pvt_sip allowtransfer was not being set to that of the peer's setting.
Therefore, the global allowtransfer setting was being used instead which would
lead to calls not being transfered if the global setting was set to 'no' despite
the setting on the peer being 'yes' and vice versa, calls would be allowed to
transfer even if the peer's setting was 'no' but the global setting was 'yes'.
(Closes issue ASTERISK-19856)
Reported by: Jacek
Tested by: Michael L. Young, Jacek
Patches:
issue-asterisk-19856-branch10-v3.diff uploaded by
Michael L. Young (license 5026)
Merged revisions 367730 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r367782 | rmudgett | 2012-05-25 11:30:55 -0500 (Fri, 25 May 2012) | 18 lines
AST-2012-007: Fix IAX receiving HOLD without suggested MOH class crash.
* Made schedule_delivery() set the received frame f->data.ptr to NULL if
the datalen is zero.
* Fix queue_signalling() memcpy() size error.
* Made queue_signalling() not use C++ keyword variable names.
(closes issue ASTERISK-19597)
Reported by: mgrobecker
Patches:
jira_asterisk_19597_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Michael L. Young
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Merged revisions 367781 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r367844 | mjordan | 2012-05-29 13:33:20 -0500 (Tue, 29 May 2012) | 21 lines
AST-2012-008: Fix remote crash vulnerability in chan_skinny
When a skinny session is unregistered, the corresponding device pointer is set
to NULL in the channel private data. If the client was not in the on-hook state
at the time the connection was closed, the device pointer can later be
dereferened if a message or channel event attempts to use a line's pointer to
said device.
The patches prevent this from occurring by checking the line's pointer in
message handlers and channel callbacks that can fire after an unregistration
attempt.
(closes issue ASTERISK-19905)
Reported by: Christoph Hebeisen
Tested by: mjordan, Damien Wedhorn
Patches:
AST-2012-008-1.8.diff uploaded by mjordan (license 6283)
AST-2012-008-10.diff uploaded by mjordan (licesen 6283)
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r367907 | rmudgett | 2012-05-29 17:28:55 -0500 (Tue, 29 May 2012) | 17 lines
Coverity Report: Fix issues for error type REVERSE_INULL (deprecated modules)
* Fix only issue pointed out by deprecated_REVERSE_INULL.txt for
app_meetme.c in find_user().
* Change use of %i to %d in sscanf() in find_user(). The use of %i gives
unexpected parsing because it can accept hex, octal, and decimal integer
formats.
* Changed other uses of %i in app_meetme() to use %d for consistency.
(issue ASTERISK-19648)
Reported by: Matt Jordan
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Merged revisions 367906 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r367978 | rmudgett | 2012-05-30 12:39:24 -0500 (Wed, 30 May 2012) | 19 lines
Fix deadlock when executing CLI "pri show channels" and "ss7 show channels" commands.
* Fix sig_pri_lock_owner() to avoid deadlock properly.
* Code pri_grab() better.
* Fix sig_ss7_lock_owner() to avoid deadlock properly.
* Code ss7_grab() better.
(closes issue ASTERISK-19854)
Reported by: Jaxon
Patches:
jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded by rmudgett (Modified to do the same thing to sig_ss7)
Tested by: Jaxon
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Merged revisions 367976 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r367981 | rmudgett | 2012-05-30 13:07:28 -0500 (Wed, 30 May 2012) | 7 lines
Use the DEADLOCK_AVOIDANCE() macro instead.
(issue ASTERISK-19854)
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Merged revisions 367980 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r368042 | rmudgett | 2012-05-31 13:20:15 -0500 (Thu, 31 May 2012) | 10 lines
Coverity Report: Fix issues for error type REVERSE_INULL (core modules)
* Fixes findings: 0-2,5,7-15,24-26,28-31
(issue ASTERISK-19648)
Reported by: Matt Jordan
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Merged revisions 368039 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r368093 | elguero | 2012-05-31 22:28:09 -0500 (Thu, 31 May 2012) | 17 lines
Add documentation to function CHANNEL for options echocan_mode and buffers
The ability to set "echocan_mode" and "buffers" through the dialplan was added
to chan_dahdi some time ago. This patch adds some documentation to
func_channel.
(Closes issue ASTERISK-19911)
Reported by: Dale Noll
Tested by: Michael L. Young
Patches:
asterisk-19911-branch18.diff uploaded by Michael L. Young (license 5026)
Merged revisions 368092 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r368267 | kpfleming | 2012-06-01 15:22:44 -0500 (Fri, 01 Jun 2012) | 20 lines
Improve SDP parsing warning messages
* 'Unsupported media type' is only reported when that is in fact the case,
not when a supported media type is included in an 'm' line that has an
invalid format.
* All warning messages related to parsing 'm' lines now include the 'm' line contents.
* (minor bugfix) newline added to port-number-zero warning messages.
* Warning messages improved to use RFC-specified terminology for various items.
* Warnings for offers that include more than one port for a single media type now
include the media type.
Merged revisions 368308 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r368407 | rmudgett | 2012-06-04 14:08:52 -0500 (Mon, 04 Jun 2012) | 23 lines
Fix potential deadlock between masquerade and chan_local.
* Restructure ast_do_masquerade() to not hold channel locks while it calls
ast_indicate().
* Simplify many calls to ast_do_masquerade() since it will never return a
failure now. If it does fail internally because a channel driver callback
operation failed, the only thing ast_do_masquerade() can do is generate a
warning message about strange things may happen and press on.
* Fixed the call to ast_bridged_channel() in ast_do_masquerade(). This
change fixes half of the deadlock reported in ASTERISK-19801 between
masquerades and chan_iax.
Merged revisions 368469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r368499 | mmichelson | 2012-06-04 17:02:26 -0500 (Mon, 04 Jun 2012) | 16 lines
Relay proper SIP responses on calling side.
Revision 351130 broke corect HANGUPCAUSE setting
for the 404 case in chan_sip. Other cases were also
potentially broken. This patch fixes the relaying
of causes to be what they used to be.
(closes issue ASTERISK-19914)
Reported by Pavel Troller
Tested by Walter Doekes (via a reviewboard test to be committed later)
Patches:
chan_sip.diff uploaded by Pavel Troller (license #6302)
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Merged revisions 368498 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r368524 | kmoore | 2012-06-05 10:19:58 -0500 (Tue, 05 Jun 2012) | 11 lines
Ensure that pages and emails are sent using RFC822-compliant date format
When localization was added to app_voicemail, these headers were altered
when they should have remained in en_US format for RFC compliance. This
reverts the changes to those two lines.
(closes issue ASTERISK-19876)
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Merged revisions 368520 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r368536 | kmoore | 2012-06-05 10:27:01 -0500 (Tue, 05 Jun 2012) | 8 lines
Resolve some build warnings
My newly upgraded compiler caught these usages of uninitialized values.
They weren't actually used.
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Merged revisions 368533 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r368568 | rmudgett | 2012-06-05 20:10:10 -0500 (Tue, 05 Jun 2012) | 15 lines
Fix parked call performing a DTMF blind transfer after being retrieved.
When a parked call was retrieved from the parking lot, it could not do a
blind transfer because it caused the involved calls to be hung up
unconditionally.
* Made the ParkedCall application return the ast_bridge_call() return
value.
Merged revisions 368586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 365155,365160,365299,365320,365399,365475,365478,365575,365632,365701,365898,365990,366049,366053,366106,366168,366241,366297,366390,366412,366591,366598,366741,366792,366881,366884,366948,367003,367028,367267,367299,367369,367417,367470,367562,367679,367731,367782,367844,367907,367978,367981,368042,368093,368267,368310,368407,368470,368499,368524,368536,368568,368587 from http://svn.asterisk.org/svn/asterisk/branches/10
Matthew Jordan [Wed, 6 Jun 2012 17:35:21 +0000 (17:35 +0000)]
Add feature modifier to versions produced from branches
Certain branches, such as Certified Asterisk, may have a modifier added to
them that specifies the features available in that branch. For branches, this
modifier is expected to be reflected in the location of the branch in
subversion. For example, a subversion of URL of /certified/branches/1.8.11
would have a feature modifier of 'certified'. This is slightly different then
how features are determined for tags, where the feature is part of the actual
tag name, e.g., "10.5.0-digiumphones".
In keeping with the nomenclature used for tags, the feature specifier for
branches is translated and placed after the revision numbers. For the example
given previously, this would result in a branch version of
"Asterisk SVN-branch-1.8.11-cert-rXXXXXX".
Mark Michelson [Mon, 21 May 2012 19:16:37 +0000 (19:16 +0000)]
Add "send to voicemail" Digium phone functionality to Asterisk.
This change accommodates two methods by which calls can be directed to
a user's voicemail.
* Incoming calls can be redirected to any user's voicemail.
* Established calls can be blind transferred to any user's voicemail.
Digium phones indicate the desire to direct a call to voicemail by using
a Diversion header with a reason parameter of "send_to_vm".
This patch adds the "send_to_vm" reason as a valid redirecting reason. In
addition, chan_sip.c has been modified to update redirecting information
on the transferred channel by reading a Diversion header on a REFER request.
(closes issue AST-871)
Reported by Malcolm Davenport
projects / thirdparty / asterisk.git / log
