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.
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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.
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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)
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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.
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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.
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This patch addresses a number of memory leaks in a variety of modules that were
found by a static analysis tool. A brief summary of the changes:
* app_minivm: free ast_str objects on off nominal paths
* app_page: free the ast_dial object if the requested channel technology
cannot be appended to the dialing structure
* app_queue: if a penalty rule failed to match any existing rule list
names, the created rule would not be inserted and its memory
would be leaked
* app_read: dispose of the created silence detector in the presence of
off nominal circumstances
* app_voicemail: dispose of an allocated unique ID field for MWI event
un-subscribe requests in off nominal paths; dispose of
configuration objects when using the secret.conf option
* chan_dahdi: dispose of the allocated frame produced by ast_dsp_process
* chan_iax2: properly unref peer in CLI command "iax2 unregister"
* chan_sip: dispose of the allocated frame produced by sip_rtp_read's
call of ast_dsp_process; free memory in parse unit tests
* func_dialgroup: properly deref ao2 object grhead in nominal path of
dialgroup_read
* func_odbc: free resultset in off nominal paths of odbc_read
* cli: free match_list in off nominal paths of CLI match completion
* config: free comment_buffer/list_buffer when configuration file load
is unchanged; free the same buffers any time they were
created and config files were processed
* data: free XML nodes in various places
* enum: free context buffer in off nominal paths
* features: free ast_call_feature in off nominal paths of applicationmap
config processing
* netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct
that is allocated by the method. Failures in
ast_sockaddr_resolve could result in the users of the method
not knowing whether or not the buffer was allocated. The
method will now not allocate the ast_sockaddr struct if it
will return failure.
* pbx: cleanup hash table traversals in off nominal paths; free
ignore pattern buffer if it already exists for the specified
context
* xmldoc: cleanup various nodes when we no longer need them
* main/editline: various cleanup of pointers not being freed before being
assigned to other memory, cleanup along off nominal paths
* menuselect/mxml: cleanup of value buffer for an attribute when that attribute
did not specify a value
* res_calendar*: responses are allocated via the various *_request method
returns and should not be allocated in the various
write_event methods; ensure attendee buffer is freed if no
data exists in the parsed node; ensure that calendar objects
are de-ref'd appropriately
* res_jabber: free buffer in off nominal path
* res_musiconhold: close the DIR* object in off nominal paths
* res_rtp_asterisk: if we run out of ports, close the rtp socket object and free
the rtp object
* res_srtp: if we fail to create the session in libsrtp, destroy the
temporary ast_srtp object
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.
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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
Fix MusicOnHold in MeetMe so that it always uses the class if it's been defined
There were a few instances of restarting music on hold in meetme that would cause
Asterisk to revert to the default class of music on hold for no adequate reason.
The error message for failure to stop autoservice after a gosub or macro call
during a dial was removed for macro while Asterisk 1.4 was still being actively
developed. The corresponding gosub error message was never removed.
Don't add an empty MESSAGE_DATA(key) header if it doesn't already exist.
Doing Set(MESSAGE_DATA(key)=) would add an empty key header if the key
header did not already exist. If it already existed it would delete it.
* Made msg_set_var_full() exit early if the named variable did not already
exist and the value to set is empty.
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r361560 | mjordan | 2012-04-06 15:32:13 -0500 (Fri, 06 Apr 2012) | 13 lines
Fix memory leak when using MeetMeAdmin 'e' option with user specified
A memory leak/reference counting leak occurs if the MeetMeAdmin 'e' command
(eject last user that joined) is used in conjunction with a specified user.
Regardless of the command being executed, if a user is specified for the
command, MeetMeAdmin will look up that user. Because the 'e' option kicks
the last user that joined, as opposed to the one specified, the reference to
the user specified by the command would be leaked when the user variable
was assigned to the last user that joined.
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Fix memory leak in res_calendar_ews when event email address node is empty
If the XML calendar data returned by a Microsoft Exchange Web Service
specifies an XML Event E-Mail Address ("EmailAddress"), and no e-mail address
is provided, a condition existed where an ast_calendar_attendee struct would
be allocated but not appended to the list of attendees. Because of that,
the memory associated with the attendee would never be freed. This patch
frees the memory if no e-mail address is provided.
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Change SHARED function to use a safe traversal when modifying a variable
When the SHARED function modifies a variable, it removes it from its list of
variables and reinserts the new value at the head of the list of variables.
Doing this inside a standard list traversal can be dangerous, as the
standard list traversal does not account for the list being changed. While
the code in question should not cause a use after free violation due to its
breaking out of the loop after freeing the variable, it could lead to a
maintenance issue if the loop was modified. This also fixes a violation
reported by a static analysis tool, which also makes this code easier to
maintain in the future.
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Prevent invalid access of free'd memory if DAHDI channel during an MWI event
In the MWI processing loop, when a valid event occurs the temporary caller ID
information is deallocated. If a new DAHDI channel is successfully created,
the event is passed up to the analog_ss_thread without error and the loop
exits. If, however, the DAHDI channel is not created, then the caller ID
struct has been free'd, and the gains reset to their previous level. This
will almost certainly cause an invalid access to the free'd memory, either
in subsequent calls to callerid_free or calls to callerid_feed.
This patch makes it so that we only free the caller ID structure if a
DAHDI channel is successfully created, and we bump the gains back up
if we fail to make a DAHDI channel.
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Allow func_curl to exit gracefully if list allocation fails during write
If the global_curl_info data structure could not be allocated, the
datastore associated with the operation would be free'd, but the function
would not return. This would later dereference the datastore, almost
certainly causing Asterisk to crash. With this patch, if the data
structure is not allocated the method will return an error code, and
not attempt any further operation.
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Fix crash caused by unloading or reloading of res_http_post
When unlinking itself from the registered HTTP URIs, res_http_post could
inadvertently free all URIs registered with the HTTP server. This patch
modifies the unregister method to only free the URI that is actually
being unregistered, as opposed to all of them.
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Prevent invalid access of free'd memory if DAHDI channel during an MWI event
In the MWI processing loop, when a valid event occurs the temporary caller ID
information is deallocated. If a new DAHDI channel is successfully created,
the event is passed up to the analog_ss_thread without error and the loop
exits. If, however, the DAHDI channel is not created, then the caller ID
struct has been free'd, and the gains reset to their previous level. This
will almost certainly cause an invalid access to the free'd memory, either
in subsequent calls to callerid_free or calls to callerid_feed.
* Rework the -r361705 patch to better manage the cs and mtd allocated
resources.
* Fixed use of mwimonitoractive flag to be correct if the mwi_thread()
fails to start.
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Change default value of 'ignorebusy' on Queue members so that behavior is more like 1.8
Prior to this patch, in order to restore that behavior, a function would have
to be used on the QueueMember to make the ringinuse option do anything, which
is pretty unreasonable.
This change to the build system rips out any usage of PROC along with
architecture-specific optimizations in favor of using -march=native where it is
supported. This fixes broken builds on 64bit Intel systems and results in
better optimized code on systems running GCC 4.2+.
Previously, setting trunkfreq had no effect on initial load or on reload and
only ever used the default value. This causes trunkfreq to be used
appropriately on initial load and reload.
Make ForkCDR e option not set end time of the newly forked CDR log
Prior to this patch, ForkCDR's e option would immediately set the end time of the forked
CDR to that of the CDR that is being terminated. This resulted in the new CDR's end time
being roughly the same as it's beginning time (which is in turn roughly the same as the
original's end time).
(closes issue ASTERISK-19164)
Reported by: Steve Davies
Patches:
cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012)
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Check for IO stream failures in various format's truncate/seek operations
For the formats that support seek and/or truncate operations, many of
the C library calls used to determine or set the current position indicator
in the file stream were not being checked. In some situations, if an error
occurred, a negative value would be returned from the library call. This
could then be interpreted inappropriately as positional data.
This patch checks the return values from these library calls before
using them in subsequent operations.
Fix handling of negative return code when storing voicemails in ODBC storage
When storing a voicemail message using an ODBC connection to a database, the
voicemail message is first stored on disk. The sound file associated with
the message is read into memory before being transmitted to the database.
When this occurs, a failure in the C library's lseek function would cause a
negative value to be passed to the mmap as the size of the memory map to
create. This would almost certainly cause the creation of the memory map to
fail, resulting in the message being lost.
In chan_agent, while handling a channel indicate, the agent channel driver
must obtain a lock on both the agent channel, as well as the channel the
agent channel is using. To do so, it attempts to lock the other channel
first, then unlock the agent channel which is locked prior to entry into
the indicate handler. If this unlock fails with a negative return value,
which can occur if the object passed to agent_indicate is an invalid ao2
object or is NULL, the return value is passed directly to strerror, which
can only accept positive integer values.
In chan_dahdi, the return value of dahdi_get_index is used to directly
index into the sub-channel array. If dahd_get_index returns a negative
value, it would use that value to index into the array, which could cause
an invalid memory access. If dahdi_get_index returns a negative number,
we now default to SUB_REAL.
Turn off warning message when bind address is set to any.
When a bind address is set to an ANY address (udpbindport=::), a warning message
is displayed stating that "Address remapping activated in sip.conf but we're
using IPv6, which doesn't need it. Please remove 'localnet' and/or 'externaddr'
settings." But if one is running dual stack, we shouldn't be told to turn those
settings off.
This patch checks if the bind address is an ANY address or not. The warning
message will now only be displayed if the bind address is NOT an ANY address and
IPv6 is being used.
Also, updated the copyright year.
(closes issue ASTERISK-19456)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
chan_sip_ipv6_message.diff uploaded by Michael L. Young (license 5026)
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Fix error that caused seek format operations to set max file size to '1' or '0'
A very inappropriate placement of a ')' (introduced in r362151) caused the
maximum size of a file to be set as the result of a comparison operation, as
opposed to the result of the ftello operation. This resulted in seeking being
restricted to the beginning of the file, or 1 byte into the file. Thanks to
the Asterisk Test Suite for properly freaking out about this on at least one
test.
(issue ASTERISK-19655)
Reported by: Matt Jordan
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Fix places where a negative return from ftello could be used as invalid input
In a variety of locations in both reading and writing a file, the result
from the C library function ftello is used as input to other functions. For
the parameters and functions in question, a negative value is invalid input.
This patch checks the return value from the ftello function to determine if
we were able to determine the current position in the file stream and, if not,
fail gracefully.
Make use of va_args more appropriate to form in various res_config modules plus utils.
A number of va_copy operations weren't matched with a corresponding va_end in res_config_odbc. Also, there was a potential for va_end to be invoked twice on the same va_arg in utils, which would mean invoking va_end on an undefined variable... which is bad.
va_end is removed from various functions in config_pgsql and config_curl since they aren't making their own copy. The invokers of those functions are responsible for calling va_end on them.
(issue ASTERISK-19451)
Reported by: Walter Doekes
Review: https://reviewboard.asterisk.org/r/1848/
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Fix places in main where a negative return value could impact execution
This patch addresses a number of modules in main that did not handle the
negative return value from function calls adequately, or were not sufficiently
clear that the conditions leading to improper handling of the return values
could not occur. This includes:
* asterisk.c: A negative return value from the read function would be used
directly as an index into a buffer. We now check for success of the read
function prior to using its result as an index.
* manager.c: Check for failures in mkstemp and lseek when handling the
temporary file created for processing data returned from a CLI command in
action_command. Also check that the result of an lseek is sanitized prior
to using it as the size of a memory map to allocate.
Fix places in resources where a negative return value could impact execution
This patch addresses a number of modules in resources that did not handle the
negative return value from function calls adequately. This includes:
* res_agi.c: if the result of the read function is a negative number,
indicating some failure, the result would instead be treated as the number
of bytes read. This patch now treats negative results in the same manner
as an end of file condition, with the exception that it also logs the
error code indicated by the return.
* res_musiconhold.c: if spawn_mp3 fails to assign a file descriptor to srcfd,
and instead assigns a negative value, that file descriptor could later be
passed to functions that require a valid file descriptor. If spawn_mp3 fails,
we now immediately retry instead of continuing in the logic.
* res_rtp_asterisk.c: if no codec can be matched between two RTP instances
in a peer to peer bridge, we immediately return instead of attempting to
use the codec payload type as an index to determine the appropriate negotiated
codec.
Handle case where an unknown format is used to get the preferred codec size
In ast_codec_pref_getsize, if an unknown format is passed to the method,
no preferred codec will be selected and a negative number will be used to
index into the format list. The method now logs an unknown format as a
warning, and returns an empty format list.
Add ability to ignore layer 1 alarms for BRI PTMP lines.
Several telcos bring the BRI PTMP layer 1 down when the line is idle.
When layer 1 goes down, Asterisk cannot make outgoing calls. Incoming
calls could fail as well because the alarm processing is handled by a
different code path than the Q.931 messages.
* Add the layer1_presence configuration option to ignore layer 1 alarms
when the telco brings layer 1 down. This option can be configured by span
while the similar DAHDI driver teignorered=1 option is system wide. This
option unlike layer2_persistence does not require libpri v1.4.13 or newer.
* chan_mobile: Fixed an overrun where the cind_state buffer (an integer array
of size 16) would be overrun due to improper bounds checking. At worst, the
buffer can be overrun by a total of 48 bytes (assuming 4-byte integers),
which would still leave it within the allocated memory of struct hfp. This
would corrupt other elements in that struct but not necessarily cause any
further issues.
* app_sms: The array imsg is of size 250, while the array (ud) that the data
is copied into is of size 160. If the size of the inbound message is
greater then 160, up to 90 bytes could be overrun in ud. This would corrupt
the user data header (array udh) adjacent to ud.
* chan_unistim: A number of invalid memmoves are corrected. These would move
data (which may or may not be valid) into the ends of these buffers.
* asterisk: ast_console_toggle_loglevel does not check that the console log
level being set is less then or equal to the allowed log levels of 32.
* format_pref: In ast_codec_pref_prepend, if any occurrence of the specified
codec is not found, the value used to index into the array pref->order
would be one greater then the maximum size of the array.
* jitterbuf: If the element being placed into the jitter buffer lands in the
last available slot in the jitter history buffer, the insertion sort attempts
to move the last entry in the buffer into one slot past the maximum length
of the buffer. Note that this occurred for both the min and max jitter
history buffers.
* tdd: If a read from fsk_serial returns a character that is greater then 32,
an attempt to read past one of the statically defined arrays containing the
values that character maps to would occur.
* localtime: struct ast_time and tm are not the same size - ast_time is larger,
although it contains the elements of tm within it in the same layout. Hence,
when using memcpy to copy the contents of tm into ast_time, the size of tm
should be used, as opposed to the size of ast_time.
* extconf: this treats ast_timing's minmask array as if it had a length of 48,
when it has defined the size of the array as 24. pbx.h defines minmask as
having a size of 48.
(issue ASTERISK-19668)
Reported by: Matt Jordan
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Handle multiple commands per connection via netconsole
Asterisk would accept multiple NULL-delimited CLI commands via the
netconsole socket, but would occasionally miss a command due to the
command not being completely read into the buffer. This patch ensures
that any partial commands get moved to the front of the read buffer,
appended to, and properly sent.
Prevent a crash in ExternalIVR when the 'S' command is sent first.
If the first command sent from an ExternalIVR client is an 'S' command, we were
blindly removing the first element from the play list and deferencing it, even
if it was NULL. This corrects that and also locks appropriately in one place.
(issue ASTERISK-17889)
Reported by: Chris Maciejewski
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A couple of unit tests did not have have leading or trailing backslashes when
setting their test category resulting in a warning message being displayed.
Added the backslash where needed.
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Document Speech* apps hangup on failure and suggest TryExec
The Speech API apps return -1 on failure, which will hang up the channel. This
may not be desirable behavior for some, but it isn't something that can be
changed without breaking people's dialplans or writing an option to all of the
Speech apps that does what TryExec already does. This patch documents the
hangup behavior of the apps, and suggests TryExec as the solution.
The Security Events Framework API was changed while adding the generation of
security events in chan_sip. A payload type and name was missed from being
added to struct ie_maps.
(closes issue ASTERISK-19759)
Reported by: Michael L. Young
Patches:
issue-asterisk-19759.diff uploaded by Michael L. Young (license 5026)
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r362998 | rmudgett | 2012-04-20 20:45:13 -0500 (Fri, 20 Apr 2012) | 5 lines
Update app_dial M and U option GOTO return value documentation.
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Fix recalled party B feature flags for a failed DTMF atxfer.
1) B calls A with Dial option T
2) B DTMF atxfer to C
3) B hangs up
4) C does not answer
5) B is called back
6) B answers
7) B cannot initiate transfers anymore
* Add dial features datastore to recalled party B channel that is a copy
of the original party B channel's dial features datastore.
* Extracted add_features_datastore() from add_features_datastores().
* Renamed struct ast_dial_features features_caller and features_callee
members to my_features and peer_features respectively. These better names
eliminate the need for some explanatory comments.
* Simplified code accessing the struct ast_dial_features datastore.
Clear ISDN channel resetting state if the peer continues to use it.
Some ISDN switches occasionally fail to send a RESTART ACKNOWLEDGE in
response to a RESTART request.
* Made the second SETUP received after sending a RESTART request clear the
channel resetting state as if the peer had sent the expected RESTART
ACKNOWLEDGE before continuing to process the SETUP. The peer may not be
sending the expected RESTART ACKNOWLEDGE.
Make DAHDISendCallreroutingFacility wait 5 seconds for a reply before disconnecting the call.
Some switches may not handle the call-deflection/call-rerouting message if
the call is disconnected too soon after being sent. Asteisk was not
waiting for any reply before disconnecting the call.
* Added a 5 second delay before disconnecting the call to wait for a
potential response if the peer does not disconnect first.
(closes issue ASTERISK-19708)
Reported by: mehdi Shirazi
Patches:
jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett
........
The reference held for SIP blind transfers using the Replaces header in an
INVITE was never freed on success and also failed to be freed in some error
conditions. This caused a file descriptor leak since the RTP structures in use
at the time of the transfer were never freed. This reference leak and another
relating to subscriptions in the same code path have now been corrected.
(Closes issue ASTERISK-19579)
Reported by: Maciej Krajewski
Tested by: Maciej Karjewski
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Fix DTMF atxfer running h exten after the wrong bridge ends.
When party B does an attended transfer of party A to party C, the
attending bridge between party B and C should not be running an h exten
when the bridge ends. Running an h exten now sets a softhangup flag to
ensure that an AGI will run in dead AGI mode.
* Set the AST_FLAG_BRIDGE_HANGUP_DONT on the party B channel for the
attending bridge between party B and C.
fix a wrong behavior of alarm timezones in caldav and icalendar when an alarm doesnt use utc. This change uses the same timezone from the start time.
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r364204 | mjordan | 2012-04-27 09:44:13 -0500 (Fri, 27 Apr 2012) | 23 lines
Allow for reloading SRTP crypto keys within the same SIP dialog
As a continuation of the patch in r356604, which allowed for the
reloading of SRTP keys in re-INVITE transfer scenarios, this patch
addresses the more common case where a new key is requested within
the context of a current SIP dialog. This can occur, for example, when
certain phones request a SIP hold.
Previously, once a dialog was associated with an SRTP object, any
subsequent attempt to process crypto keys in any SDP offer - either
the current one or a new offer in a new SIP request - were ignored. This
patch changes this behavior to only ignore subsequent crypto keys within
the current SDP offer, but allows future SDP offers to change the keys.
(issue ASTERISK-19253)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Allow SIP pvts involved in Replaces transfers to fall out of reference sooner
Unref the SIP pvt stored in the refer structure as soon as it is no longer
needed so that the pvt and associated file descriptors can be freed sooner.
This change makes a reference decrement unnecessary in code that handles SIP
BYE/Also transfers which should not touch the reference anyway.
(Closes issue ASTERISK-19579)
Reported by: Maciej Krajewski
Tested by: Maciej Krajewski
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Prevent overflow in calculation in ast_tvdiff_ms on 32-bit machines
The method ast_tvdiff_ms attempts to calculate the difference, in milliseconds,
between two timeval structs, and return the difference in a 64-bit integer.
Unfortunately, it assumes that the long tv_sec/tv_usec members in the timeval
struct are large enough to hold the calculated values before it returns. On
64-bit machines, this might be the case, as a long may be 64-bits. On 32-bit
machines, however, a long may be less (32-bits), in which case, the calculation
can overflow.
This overflow caused significant problems in MixMonitor, which uses the method
to determine if an audio factory, which has not presented audio to an audiohook,
is merely late in providing said audio or will never provide audio. In an
overflow situation, the audiohook would incorrectly determine that an audio
factory that will never provide audio is merely late instead. This led to
situations where a MixMonitor never recorded any audio. Note that this happened
most frequently when that MixMonitor was started by the ConfBridge application
itself, or when the MixMonitor was attached to a Local channel.
(issue ASTERISK-19497)
Reported by: Ben Klang
Tested by: Ben Klang
Patches:
32-bit-time-overflow-10-2012-04-26.diff (license #6283) by mjordan
(closes issue ASTERISK-19727)
Reported by: Mark Murawski
Tested by: Michael L. Young
Patches:
32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan)
Fix ast_parse_arg numeric type range checking and add tests
ast_parse_arg wasn't checking for strto* parse errors or limiting
the results by the actual range of the numeric types. This patch fixes
that and adds unit tests as well.
Fix configuring custom sound_leader_has_left in confbridge.conf
The configuration option to specify a custom sound_leader_has_left file for a
conference bridge was not being parsed. This patch fixes it so that a custom
sound file will now be used.
(closes issue ASTERISK-19771)
Reported by: Pawel Kuzak
Tested by: Pawel Kuzak, Michael L. Young
Patches: leaderhasleft_sound.dpatch uploaded by Pawel Kuzak (license 6380)
Another very inappropriate placement of a ')' (again introduced in r362151)
caused the various truncate operations to attempt to truncate the sound file
at a position of '0'.
Merged revisions 364769 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r364787 | kmoore | 2012-05-01 14:07:09 -0500 (Tue, 01 May 2012) | 12 lines
Play conf-placeintoconf message to the correct channel
Correct the code in app_confbridge to play the conf-placeintoconf message to
the marked user entering the bridge instead of to the conference while the
marked user hears silence.
(closes issue ASTERISK-19641) Reported-by: Mark A Walters
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Merged revisions 364786 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r364845 | rmudgett | 2012-05-01 16:50:32 -0500 (Tue, 01 May 2012) | 7 lines
* Fix error path resouce leak in local_request().
* Restructure local_request() to reduce indentation.
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Merged revisions 364840 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r364900 | mmichelson | 2012-05-01 18:10:16 -0500 (Tue, 01 May 2012) | 16 lines
Fix Coverity-reported ARRAY_VS_SINGLETON error.
As it turned out, this wasn't a huge deal. We were calling
ast_app_parse_options() for a set of options of which none
took arguments. The proper thing to do for this case is to
pass NULL for the "args" parameter here. We were instead passing
a seemingly-randomly chosen char * from the function. While this
would never get written to, you can rest assured things would
have gotten bad had new options (which took arguments) been added
to func_volume.
(closes issue ASTERISK-19656)
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Merged revisions 364899 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r364903 | rmudgett | 2012-05-01 18:14:12 -0500 (Tue, 01 May 2012) | 10 lines
Merged revisions 364902 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r364965 | mjordan | 2012-05-01 21:44:15 -0500 (Tue, 01 May 2012) | 11 lines
Only log a failure to get read/write samples from factories if it didn't happen
In audiohook_read_frame_both, anytime samples are obtained from the read/write
factories a debug statement is logged stating that samples were not obtained
from the factories. This statement used to only occur if option_debug was
turned on and no samples were obtained; in some refactoring when the
option_debug statement was removed, the "else" clause was removed as well.
This patch makes it so that those debug log statements only occur if the
condition leading up to them actually happened.
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r365014 | elguero | 2012-05-02 11:16:03 -0500 (Wed, 02 May 2012) | 18 lines
Update security events unit tests
The security events framework API was changed in Asterisk 10 but the unit tests
were not updated at the same time.
This patch does the following:
* Adds two more security events that were added to the API
* Add challenge, received_challenge and received_hash in the inval_password
security event unit test
(issue ASTERISK-19760)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
issue-asterisk-19760-branch10.diff uploaded by Michael L. Young (license 5026)
Fix a CEL LINKEDID_END race and local channel linkedids
This patch has the ;2 channel inherit the linkedid of the ;1 channel and fixes
the race condition by no longer scanning the channel list for "other" channels
with the same linkedid. Instead, cel.c has an ao2 container of linkedid strings
and uses the refcount of the string as a counter of how many channels with the
linkedid exist. Not only does this eliminate the race condition, but it also
allows us to look up the linkedid by the hashed key instead of traversing the
entire channel list.
Don't leak a ref if out of memory and can't link the linkedid
If the ao2_link fails, we are most likely out of memory and bad things
are going to happen. Before those bad things happen, make sure to clean
up the linkedid references.
This patch also adds a comment explaining why linkedid can't be passed
to both local channel allocations and combines two ao2_ref calls into 1.
Merged revisions 365006,365068 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 361208,361211,361270,361330,361333,361381,361422,361472,361522,361560,361607,361658,361706,361754,361804,361855,361907,361956,361981,362080,362084,362152,362202,362205,362264,362305,362356-362357,362360,362364,362377,362429,362496,362537,362587,362678,362681,362730,362816,362869,362918,362998,363212,363376,363429,363688,363734,363789,363876,363935,363987,364047,364065,364109,364163,364204,364259,364285,364342,364365,364369,364536,364579,364650-364651,364777,364787,364845,364900,364903,364965,365014,365083 from http://svn.asterisk.org/svn/asterisk/branches/10
Replace GNU old-style field designator extensions to fix clang warnings
(issue ASTERISK-19540)
Reported by: Makoto Dei
Patches:
clang-gnu-designator.patch uploaded by Makoto Dei (license 5027)
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Also add from the patch the portion in res_fax_spandsp that didn't apply to 1.8
AST-2012-005: Fix remotely exploitable heap overflow in keypad button handling
When handling a keypad button message event, the received digit is placed into
a fixed length buffer that acts as a queue. When a new message event is
received, the length of that buffer is not checked before placing the new digit
on the end of the queue. The situation exists where sufficient keypad button
message events would occur that would cause the buffer to be overrun. This
patch explicitly checks that there is sufficient room in the buffer before
appending a new digit.
(closes issue ASTERISK-19592)
Reported by: Russell Bryant
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Merged revisions 363100 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
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Reference skinny_subchannel object instead of skinny_device for r363103
The check-in to resolve ASTERISK-19592 (r363103) failed to switch to the
skinny_subchannel object instead of the skinny_device when attempting to
reference the buffer for the keypad digits. This patch fixes that.
AST-2012-006: Fix crash in UPDATE handling when no channel owner exists
If Asterisk receives a SIP UPDATE request after a call has been terminated and
the channel has been destroyed but before the SIP dialog has been destroyed, a
condition exists where a connected line update would be attempted on a
non-existing channel. This would cause Asterisk to crash. The patch resolves
this by first ensuring that the SIP dialog has an owning channel before
attempting a connected line update. If an UPDATE request is received and no
channel is associated with the dialog, a 481 response is sent.
(closes issue ASTERISK-19770)
Reported by: Thomas Arimont
Tested by: Matt Jordan
Patches:
ASTERISK-19278-2012-04-16.diff uploaded by Matt Jordan (license 6283)
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AST-2012-004: Fix an error that allows AMI users to run shell commands sans authorization.
As detailed in the advisory, AMI users without write authorization for SYSTEM class AMI
actions were able to run system commands by going through other AMI commands which did
not require that authorization. Specifically, GetVar and Status allowed users to do this
by setting their variable/s options to the SHELL or EVAL functions.
Also, within 1.8, 10, and trunk there was a similar flaw with the Originate action that
allowed users with originate permission to run MixMonitor and supply a shell command
in the Data argument. That flaw is fixed in those versions of this patch.
(closes issue ASTERISK-17465)
Reported By: David Woolley
Patches:
162_ami_readfunc_security_r2.diff uploaded by jrose (license 6182)
18_ami_readfunc_security_r2.diff uploaded by jrose (license 6182)
10_ami_readfunc_security_r2.diff uploaded by jrose (license 6182)
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Merged revisions 363117 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
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Revert improved identities sent in dialog-info NOTIFY requests in r360862
Revision 360862 was intended to improve identities sent in dialog-info
NOTIFY requests. Some users reported that hint became broken once this
was done. It's not clear exactly what part of the patch has caused this
regression, but broken hints are bad.
For now, this revision is being reverted so that the next releases of
Asterisk do not have bad behavior in them. The original reported issue
will have to be fixed differently in the next version of Asterisk.
(issue ASTERISK-16735)
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Merged revisions 364706 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 361143,363103-363104,363107,363156,364707 from http://svn.asterisk.org/svn/asterisk/branches/10
Kinsey Moore [Tue, 3 Apr 2012 20:08:47 +0000 (20:08 +0000)]
Fix the display of documentation for Transfer
This came up while fixing documentation generation for many other cases where
the argument separator was not being displayed properly. Now that it is
displayed properly, it shows up in the wrong place for Transfer since the '/'
is only required if Tech is present.
(related to issue ASTERISK-18168)
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Merged revisions 361040 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Mon, 2 Apr 2012 22:22:07 +0000 (22:22 +0000)]
Stop sending out RTCP if RTP is inactive
This change prevents Asterisk from sending RTCP receiver reports during a
remote bridge since it is no longer receiving media and should not be
reporting anything.
(related to ASTERISK-19366)
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Merged revisions 360987 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Thu, 29 Mar 2012 23:33:59 +0000 (23:33 +0000)]
Fix potential race condition during call pickup.
Prior to this patch, a connected line update was queued during
call pickup and then an answer frame was queued. The original
caller would presumably then have his connected line updated
and then the call would be answered.
In actuality, the answer frame was not how the call ended up
being answered. Rather, an odd section in app_dial that checks
if the called channel's state is up.
The result is that the order of the connected line update and
the answer were variable. In most cases, this wasn't actually
a bad thing. However, if the 'I' option was passed to dial, the
connected line update would be inhibited.
The fix is to queued the connected line after the answer frame is
queued. This way the race in app_dial is between two
conditions resulting in an answer. This way the connected line
update occurs after the answer every time.
(closes issue ASTERISK-19183)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Mark Michelson
Patches:
ASTERISK-19183.patch uploaded by Mark Michelson (license 5049)
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Merged revisions 360884 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Thu, 29 Mar 2012 23:05:42 +0000 (23:05 +0000)]
Improve accuracy of identifying information sent in dialog-info SIP NOTIFY requests.
This change makes use of connected party information in addition to caller ID in order
to populate local and remote XML elements in the dialog-info NOTIFYs.
(closes issue ASTERISK-16735)
Reported by: Maciej Krajewski
Tested by: Maciej Krajewski
Patches:
local_remote_hint2.diff uploaded by Mark Michelson (license 5049)
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Merged revisions 360862 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Tue, 27 Mar 2012 18:23:11 +0000 (18:23 +0000)]
Make a debug message regarding subscription changes more accurate.
I was getting confused during some testing why Asterisk was saying that
a subscription was being added when it was clearly being removed. This
fixes that confusion.
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Merged revisions 360625 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Paul Belanger [Mon, 26 Mar 2012 18:26:51 +0000 (18:26 +0000)]
Increase verbosity level for ast_verb messages
While this does not fix the issue of the CLI being flooded by 'doing
dnsmgr_lookup' messages, increasing the verbosity level above 5 should help
minimize it.
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Merged revisions 360471 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Russell Bryant [Sat, 24 Mar 2012 03:10:22 +0000 (03:10 +0000)]
app_page: Fix a memory leak on every Page().
dial_list is a dynamically allocated array that is allocated at the beginning
of Page() based on how many devices will be dialed. This was never being freed.
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Merged revisions 360363 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Russell Bryant [Sat, 24 Mar 2012 02:38:59 +0000 (02:38 +0000)]
Multiple revisions 360356-360357
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r360356 | russell | 2012-03-23 22:33:36 -0400 (Fri, 23 Mar 2012) | 6 lines
expression parser: Fix (theoretical) memory leak.
Fix a memory leak that is very unlikely to actually happen. If a malloc()
succeeded, but the following strdup() failed, the memory from the original
malloc() would be leaked.
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r360357 | russell | 2012-03-23 22:34:39 -0400 (Fri, 23 Mar 2012) | 6 lines
Rebuild parsers.
This is needed to include the last fix to main/ast_expr2.y. The changes look
much bigger as this regeneration of the code was done with newer versions of
flex and bison.
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Merged revisions 360356-360357 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Sat, 24 Mar 2012 00:37:13 +0000 (00:37 +0000)]
Make number not available presentation also set screening to network provided.
Q.951 indicates that when the presentation indicator is "Number not
available due to interworking" for a number then the screening indicator
field should be "Network provided".
* Made ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE
when the presentation is "Number not available due to interworking". This
fix makes Asterisk consistent and it also makes it consistent with earlier
branches as far as this presentation value is concerned.
* Made pri_to_ast_presentation() and ast_to_pri_presentation() conversions
handle the "Number not available due to interworking" case better in
sig_pri.c. This change is possible because the minimum required libpri
version (v1.4.11) has the necessary defines in libpri.h.
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Merged revisions 360309 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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