Matthew Jordan [Fri, 2 Nov 2012 14:50:39 +0000 (14:50 +0000)]
Help mitigate potential reinvite glare scenarios.
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
........
Merged revisions 370618 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Fri, 2 Nov 2012 14:33:35 +0000 (14:33 +0000)]
Resolve memory leaks in TLS initialization and TLS client connections
This patch resolves two sources of memory leaks when using TLS in Asterisk:
1) It removes improper initialization (and multiple re-initializations) of
portions of the SSL library. Asterisk calls SSL_library_init and
SSL_load_error_strings during SSL initialization; collectively this
obviates the need for calling any of the following during initialization
or client connection handling:
* ERR_load_crypto_strings (handled by SSL_load_error_strings)
* OpenSSL_add_all_algorithms (synonym for SSL_library_init)
* SSLeay_add_ssl_algorithms (synonym for SSL_library_init)
2) Failure to completely clean up all memory allocated by Asterisk and by
the SSL library for TLS clients. This included not freeing the SSL_CTX
object in the SIP channel driver, as well as not clearing the error
stack when the TLS client exited.
Note that these memory leaks were found by Thomas Arimont, and this patch
was essentially written by him with some minor tweaks.
(closes issue AST-889)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
patches:
(bugAST-889.patch) by Thomas Arimont (license 5525)
Matthew Jordan [Thu, 11 Oct 2012 15:46:58 +0000 (15:46 +0000)]
Fix incorrect billing duration reported when batch mode is enabled
Similar to r369351, the billing duration can be skewed when batch mode is
enabled. This happened much more rarely than the duration, as it only
occured when the call was answered (thereby indicating an actual answer
time) and immediately hung up on (indicating a billsec of 0). Since
a billing time of '0' can either mean that the call immediately ended
or that the CDR was improperly answered, we have to use additional information
to know whether or not we can trust the CDR billsec value. Prior to this
patch, we looked to see if we had a valid answer time. If we did, and
billsec was zero, we used the current time to calculate what billsec value
we could from the CDR being written. If batch mode is enabled, this will
incorrectly report a billsec value being much greater than the actual
duration of the call.
Instead of relying on the presence of an answer time to know whether or not
we can re-calculate the billsec for the CDR, we now also use the presence
of the CDR's end time to know if we need to re-calculate or whether we can
trust the billsec value that we have. This prevents erroneous jumps in the
billsec value, while still making sure that in the worst case, some billing
time will be calculated.
(closes issue AST-1016)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
........
Merged revisions 374843 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Wed, 10 Oct 2012 21:16:14 +0000 (21:16 +0000)]
app_queue: Made pass connected line updates from the caller to ringing queue members.
Party A calls Party B
Party B puts Party A on hold.
Party B calls a queue.
Ringing queue member D sees Party B identification.
Party B transfers Party A to the queue.
Queue member D does not get a connected line update for Party A.
Queue member D answers the call and still sees Party B information.
However, if Party A later transfers the call to Party C then queue member
D gets a connected line update for Party C.
* Made pass connected line updates from the caller to queue members while
the queue members are ringing.
(closes issue AST-1017)
Reported by: Thomas Arimont
(closes issue ABE-2886)
Reported by: Thomas Arimont
Tested by: rmudgett
........
Merged revisions 374801 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
........
Merged revisions 374802 from http://svn.asterisk.org/svn/asterisk/branches/1.8
David M. Lee [Fri, 5 Oct 2012 20:12:49 +0000 (20:12 +0000)]
Improve AMI long line error handling
In AMI's parser, when it receives a long line (> 1024 characters), it discards
that line, but continues to process the message normally.
Typically, this is not a problem because a) who has lines that long and b)
usually a discarded line results in an invalid message. But if that line is
specifying an optional field, then the message will be processed, you get a
'Response: Success', but things don't work the way you expected them to.
This patch changes the behavior when a line-too-long parse error occurs.
* Changes the log message to avoid way-too-long (and truncated anyways) log
messages
* Adds a 'parsing' status flag to Response: Success
* Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line is too long
* Responds with an appropriate error if parsing != MESSAGE_OKAY
(closes issue AST-961)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2142/
In handle_frm_te() after calling misdn_lib_send_event(bc,
EVENT_RELEASE_COMPLETE) bc is emptied, cleaned and set not in use,
although misdn_lib_send_event() already did the same. This is bad. When
it's not in use we are not allowed to touch it.
* Moved log message in front of the resulting actions and fixed it to
match the case.
Patches:
patch5_bccleanup.diff (license #6372) patch uploaded by Guenther Kelleter
chan_misdn: We must initialize cause on sending a DISCONNECT.
We must initialize cause on sending a DISCONNECT, so it is later correctly
indicated to ast_channel in case the answer (RELEASE/RELEASE_COMPLETE)
does not include one.
Patches:
patch7_hangupcause.diff (license #6372) patch uploaded by Guenther Kelleter
chan_misdn: setup_bc() is called too early for an incoming SETUP on TE.
This prevents the B channel from being setup for HDLC mode when requested
by the bearer capability and config option hdlc=yes. It violates
ETS300102 Ch.5.2.3.2: "The user, in any case, must not connect to the
channel until a CONNECT ACKNOWLEDGE message has been received."
* Call setup_bc() on receipt of CONNECT_ACKNOWLEGDE for PTMP, and on first
response to SETUP for PTP.
Patches:
abe-2881-2.diff (license #6372) patch uploaded by Guenther Kelleter
Modified.
David M. Lee [Thu, 4 Oct 2012 15:11:06 +0000 (15:11 +0000)]
Fix DBDelTree error codes for AMI, CLI and AGI
The AMI DBDelTree command will return Success/Key tree deleted successfully even
if the given key does not exist. The CLI command 'database deltree' had a
similar problem, but was saved because it actually responded with '0 database
entries removed'. AGI had a slightly different error, where it would return
success if the database was unavailable.
This came from confusion about the ast_db_deltree retval, which is -1 in the
event of a database error, or number of entries deleted (including 0 for
deleting nothing).
* Adds a Doxygen comment to process_db_keys explaining its retval
* Changed some poorly named res variables to num_deleted
* Specified specific errors when calling ast_db_deltree (database unavailable
vs. entry not found vs. success)
* Fixed similar bug in AGI database deltree, where 'Database unavailable'
results in successful result
(closes issue AST-967)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2138/
Fix incorrect MeetME conference bridge reference count decrementing and sometimes premature destruction.
When using the 'e' or 'E' option to MeetMe the configured conference bridges are loaded and examined to see
if any are empty. If no conference bridges are empty the caller is prompted to enter the number of one.
This operation left around a pointer to the last created conference bridge still containing participants.
When the caller that was not able to find any empty conference bridge hung up this pointer was disposed of
and the reference count of the conference bridge decremented. If there was only a single participant in the
conference bridge it was ultimately destroyed prematurely.
Mark Michelson [Tue, 11 Sep 2012 21:02:33 +0000 (21:02 +0000)]
Fix inability to shutdown gracefully due to an unending channel reference.
message.c makes use of a special message queue channel that exists
in thread storage. This channel never goes away due to the fact that
the taskprocessor used by message.c does not get shut down, meaning
that it never ends the thread that stores the channel.
This patch fixes the problem by shutting down the taskprocessor when
Asterisk is shut down. In addition, the thread storage has a destructor
that will release the channel reference when the taskprocessor is destroyed.
(closes issue AST-937)
Reported by Jason Parker
Patches:
AST-937.patch uploaded by Mark Michelson (License #5049)
Tested by Jason Parker
Matthew Jordan [Thu, 30 Aug 2012 18:48:52 +0000 (18:48 +0000)]
AST-2012-012: Resolve AMI User Unauthorized Shell Access through ExternalIVR
The AMI Originate action can allow a remote user to specify information that can
be used to execute shell commands on the system hosting Asterisk. This can
result in an unwanted escalation of permissions, as the Originate action, which
requires the "originate" class authorization, can be used to perform actions
that would typically require the "system" class authorization. Previous attempts
to prevent this permission escalation (AST-2011-006, AST-2012-004) have sought
to do so by inspecting the names of applications and functions passed in with
the Originate action and, if those applications/functions matched a predefined
set of values, rejecting the command if the user lacked the "system" class
authorization. As reported by IBM X-Force Research, the "ExternalIVR"
application is not listed in the predefined set of values. The solution for
this particular vulnerability is to include the "ExternalIVR" application in the
set of defined applications/functions that require "system" class authorization.
Unfortunately, the approach of inspecting fields in the Originate action against
known applications/functions has a significant flaw. The predefined set of
values can be bypassed by creative use of the Originate action or by certain
dialplan configurations, which is beyond the ability of Asterisk to analyze at
run-time. Attempting to work around these scenarios would result in severely
restricting the applications or functions and prevent their usage for legitimate
means. As such, any additional security vulnerabilities, where an
application/function that would normally require the "system" class
authorization can be executed by users with the "originate" class authorization,
will not be addressed. Instead, the README-SERIOUSLY.bestpractices.txt file has
been updated to reflect that the AMI Originate action can result in commands
requiring the "system" class authorization to be executed. Proper system
configuration can limit the impact of such scenarios.
(closes issue ASTERISK-20132)
Reported by: Zubair Ashraf of IBM X-Force Research
AST-2012-013: Resolve ACL rules being ignored during calls by some IAX2 peers
When an IAX2 call is made using the credentials of a peer defined in a dynamic
Asterisk Realtime Architecture (ARA) backend, the ACL rules for that peer are
not applied to the call attempt. This allows for a remote attacker who is aware
of a peer's credentials to bypass the ACL rules set for that peer.
This patch ensures that the ACLs are applied for all peers, regardless of their
storage mechanism.
(closes issue ASTERISK-20186)
Reported by: Alan Frisch
Tested by: mjordan, Alan Frisch
Mark Michelson [Mon, 27 Aug 2012 21:48:38 +0000 (21:48 +0000)]
Fix misleading documentation in agents.conf.sample regarding ackcall usage.
The documentation made it sound as if the DTMF acknowledgment was needed
at the time the agent logs in, rather than when the agent is called. This
is likely a relic from the days when there were multiple ways of logging
in agents.
(closes issue AST-962)
reported by Steve Pitts
........
Merged revisions 371787 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Mon, 27 Aug 2012 21:35:13 +0000 (21:35 +0000)]
Fix incorrect documentation of the MailboxStatus manager command.
The "Waiting" field was misdocumented as reporting the number of
messages waiting. In reality, it simply indicated the presence or
absence of waiting messages.
........
Merged revisions 371782 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Fri, 17 Aug 2012 16:02:50 +0000 (16:02 +0000)]
Add instrumentation to subsystem reloads
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)
........
Merged revisions 371436 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Mon, 13 Aug 2012 20:42:55 +0000 (20:42 +0000)]
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)
........
Merged revisions 371201 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Fri, 10 Aug 2012 21:42:36 +0000 (21:42 +0000)]
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
........
Merged revisions 371141 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Tue, 7 Aug 2012 15:49:48 +0000 (15:49 +0000)]
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
Jonathan Rose [Mon, 9 Jul 2012 14:38:18 +0000 (14:38 +0000)]
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.
Jonathan Rose [Fri, 6 Jul 2012 20:54:04 +0000 (20:54 +0000)]
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)
Mark Michelson [Fri, 6 Jul 2012 18:40:06 +0000 (18:40 +0000)]
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
Mark Michelson [Fri, 6 Jul 2012 15:20:11 +0000 (15:20 +0000)]
Fix bridging thread leak.
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
Kinsey Moore [Thu, 5 Jul 2012 19:01:52 +0000 (19:01 +0000)]
AST-2012-011: Resolve heap corruption issue with voicemail
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.
Review: https://reviewboard.asterisk.org/r/1994/
(closes issue ASTERISK-19923)
Reported by: Dan Delaney
Tested by: Dan Delaney, Julian Yap
Patches:
vm_alloc_fix.diff uploaded by kmoore (license 6273)
Matthew Jordan [Thu, 5 Jul 2012 17:01:52 +0000 (17:01 +0000)]
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)
Terry Wilson [Tue, 3 Jul 2012 16:58:16 +0000 (16:58 +0000)]
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.
Terry Wilson [Tue, 3 Jul 2012 14:27:02 +0000 (14:27 +0000)]
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/
Terry Wilson [Wed, 27 Jun 2012 20:58:51 +0000 (20:58 +0000)]
AST-2012-010: 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.
Review: https://reviewboard.asterisk.org/r/2009/
(closes issue ASTERISK-19992)
Reported by: Steve Davies
Tested by: Steve Davies, Terry Wilson
Matthew Jordan [Tue, 26 Jun 2012 13:21:13 +0000 (13:21 +0000)]
Fix crash in unloading of res_adsi module
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.
Matthew Jordan [Mon, 25 Jun 2012 19:24:55 +0000 (19:24 +0000)]
Tweak CDR change in r369351
As Tilghman pointed out on review 1996, the check to see if a CDR end time has
been set is sufficient to know whether or not the duration value can be used.
The check-in done for r369351 forgot to include this change.
Mark Michelson [Mon, 25 Jun 2012 19:13:31 +0000 (19:13 +0000)]
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
Matthew Jordan [Mon, 25 Jun 2012 19:12:35 +0000 (19:12 +0000)]
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
Richard Mudgett [Mon, 25 Jun 2012 15:57:28 +0000 (15:57 +0000)]
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.
Mark Michelson [Mon, 25 Jun 2012 14:18:09 +0000 (14:18 +0000)]
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)
Terry Wilson [Fri, 22 Jun 2012 19:28:04 +0000 (19:28 +0000)]
Don't crash on a guest directmedia call
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.
(closes issue ASTERISK-20040)
Reported by: Terry Wilson
Kinsey Moore [Fri, 22 Jun 2012 17:14:10 +0000 (17:14 +0000)]
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.
Michael L. Young [Wed, 20 Jun 2012 02:03:22 +0000 (02:03 +0000)]
Fix NULL pointer segfault in ast_sockaddr_parse()
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)
Mark Michelson [Tue, 19 Jun 2012 15:30:58 +0000 (15:30 +0000)]
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)
Kevin P. Fleming [Fri, 15 Jun 2012 15:56:08 +0000 (15:56 +0000)]
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.
Mark Michelson [Thu, 14 Jun 2012 15:23:10 +0000 (15:23 +0000)]
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.
Mark Michelson [Wed, 13 Jun 2012 20:59:01 +0000 (20:59 +0000)]
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)
Matthew Jordan [Wed, 13 Jun 2012 20:26:07 +0000 (20:26 +0000)]
Mark res_smdi/res_adsi as 'core' supported modules
Recently, various issues surrounding weak symbols 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 symbols.
(issue AST-900)
Reported by: Thomas Arimont
(issue AST-885)
Reported by: Denis Alberto Martinez
Matthew Jordan [Wed, 13 Jun 2012 14:27:57 +0000 (14:27 +0000)]
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.
Matthew Jordan [Tue, 12 Jun 2012 18:23:01 +0000 (18:23 +0000)]
Do not perform install on existing directories
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)
Mark Michelson [Tue, 12 Jun 2012 15:36:34 +0000 (15:36 +0000)]
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
Richard Mudgett [Wed, 6 Jun 2012 21:27:33 +0000 (21:27 +0000)]
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.
Mark Michelson [Wed, 6 Jun 2012 19:13:45 +0000 (19:13 +0000)]
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
Matthew Jordan [Wed, 6 Jun 2012 17:20:07 +0000 (17:20 +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".
Richard Mudgett [Wed, 6 Jun 2012 01:08:29 +0000 (01:08 +0000)]
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.
Kinsey Moore [Tue, 5 Jun 2012 15:15:57 +0000 (15:15 +0000)]
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.
Mark Michelson [Mon, 4 Jun 2012 21:56:05 +0000 (21:56 +0000)]
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)
Richard Mudgett [Mon, 4 Jun 2012 18:41:18 +0000 (18:41 +0000)]
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.
* '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.
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)
Richard Mudgett [Wed, 30 May 2012 17:21:43 +0000 (17:21 +0000)]
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
Richard Mudgett [Tue, 29 May 2012 22:25:21 +0000 (22:25 +0000)]
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.
Matthew Jordan [Tue, 29 May 2012 18:30:25 +0000 (18:30 +0000)]
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
dereferenced 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 (license 6283)
Richard Mudgett [Fri, 25 May 2012 16:28:04 +0000 (16:28 +0000)]
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
Michael L. Young [Fri, 25 May 2012 02:27:11 +0000 (02:27 +0000)]
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)
Richard Mudgett [Thu, 24 May 2012 22:21:18 +0000 (22:21 +0000)]
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.
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.
Matthew Jordan [Wed, 23 May 2012 13:06:08 +0000 (13:06 +0000)]
Update a peer's LastMsgsSent when the peer is notified of waiting messages
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. However, the value was still
presented when, either by AMI or CLI, a 'sip show peer [peer]' command
was executed. The output of the command would always display the erroneous
value of 32767/65535 for 'LastMsgsSent'.
This patch 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
Terry Wilson [Tue, 22 May 2012 17:14:20 +0000 (17:14 +0000)]
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.
Terry Wilson [Tue, 22 May 2012 16:14:16 +0000 (16:14 +0000)]
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/
Mark Michelson [Fri, 18 May 2012 16:53:47 +0000 (16:53 +0000)]
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.
Matthew Jordan [Fri, 18 May 2012 15:42:33 +0000 (15:42 +0000)]
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
Kinsey Moore [Fri, 18 May 2012 14:16:50 +0000 (14:16 +0000)]
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.
Matthew Jordan [Fri, 18 May 2012 13:58:23 +0000 (13:58 +0000)]
Fix a variety of memory leaks
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
projects / thirdparty / asterisk.git / log
