Matthew Jordan [Thu, 15 Mar 2012 19:11:03 +0000 (19:11 +0000)]
Fix remotely exploitable stack overflow in HTTP manager
There exists a remotely exploitable stack buffer overflow in HTTP digest
authentication handling in Asterisk. The particular method in question
is only utilized by HTTP AMI. When parsing the digest information, the
length of the string is not checked when it is copied into temporary buffers
allocated on the stack.
This patch fixes this behavior by parsing out pre-defined key/value pairs
and avoiding unnecessary copies to the stack.
(closes issue ASTERISK-19542)
Reported by: Russell Bryant
Tested by: Matt Jordan
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Mark Murawki [Thu, 15 Mar 2012 18:58:25 +0000 (18:58 +0000)]
Add options PreDial options 'b' and 'B' to app_dial
* Added 'b' and 'B' options to Dial. These options will allow you to run
last-minute dialplan on the caller and callee channels while the Dial
application is executing, but before the call is started. For example you
can use the 'b' option to run dialplan on the callee channel to get the name
of the newly created channel right away.
Review: https://reviewboard.asterisk.org/r/1229/
(closes issue: ASTERISK-19548)
Reported by: Mark Murawski
Tested by: Mark Murawski, Stefan Schmidt
Matthew Jordan [Thu, 15 Mar 2012 18:55:54 +0000 (18:55 +0000)]
Fix remotely exploitable stack overrun in Milliwatt
Milliwatt is vulnerable to a remotely exploitable stack overrun when using
the 'o' option. This occurs due to the milliwatt_generate function not
accounting for AST_FRIENDLY_OFFSET when calculating the maximum number of
samples it can put in the output buffer.
This patch resolves this issue by taking into account AST_FRIENDLY_OFFSET
when determining the maximum number of samples allowed. Note that at no
point is remote code execution possible. The data that is written into the
buffer is the pre-defined Milliwatt data, and not custom data.
(closes issue ASTERISK-19541)
Reported by: Russell Bryant
Tested by: Matt Jordan
Patches:
milliwatt_stack_overrun.rev1.txt by Russell Bryant (license 6283)
Note that this patch was written by Russell, even though Matt uploaded it
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Richard Mudgett [Thu, 15 Mar 2012 18:32:22 +0000 (18:32 +0000)]
Add missing connected line macro calls to initial dial for Dial and Queue apps.
The connected line interception macros do not get executed when the
outgoing channel is initially created and that channel's caller-id is
implicitly imported into the incoming channel's connected line data. If
you are using the interception macros, you would expect that they get run
for every change to a channel's connected line information outside of
normal dialplan execution.
Russell Bryant [Wed, 14 Mar 2012 23:29:32 +0000 (23:29 +0000)]
app_chanisavail: Fix use of uninitialized variable.
Ensure that status is set before it is used by resetting it during each loop
iteration. This could have resulted in incorrect results from this app.
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Russell Bryant [Wed, 14 Mar 2012 23:12:42 +0000 (23:12 +0000)]
udptl: Ensure fec[] in udptl_build_packet() is initialized.
Scan results indicated that this array could be used uninitialized. At a quick
look, it looks correct. In any case, initializing it is a Good Thing (tm).
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Russell Bryant [Wed, 14 Mar 2012 22:41:21 +0000 (22:41 +0000)]
app.h: Always initialize AST_DECLARE_APP_ARGS().
This patch ensures that the struct defined by AST_DECLARE_APP_ARGS() is always
fully initialized. I'm not sure if this fixes any real bugs, but it silences
a bunch of warnings from coverity, and is generally a good thing to do anyway.
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Richard Mudgett [Wed, 14 Mar 2012 22:38:29 +0000 (22:38 +0000)]
Fix deadlock potential with some ast_indicate/ast_indicate_data calls.
Calling ast_indicate()/ast_indicate_data() with the channel lock held can
result in a deadlock with a local channel because of how local channels
need to avoid deadlock.
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Matthew Jordan [Wed, 14 Mar 2012 18:56:15 +0000 (18:56 +0000)]
Add tests for main/jitterbuf.c
This patch adds unit tests for main/jitterbuf.c. This includes checking for
the following:
* Nominal insertion and retrieval of frames
* Insertion and retrieval of frames where the frames are inserted out of
order with respect to the previous frame
* Insertion and retrieval of frames where some number of frames that would
occur in the expected sequence are instead dropped
* Insertion and retrieval of frames with an arrival time that does not occur
at the same rate as the surrounding frames
* Resynchronization of the jitter buffer when an inserted frame breaks the
resynchronization threshold
* Overfilling of the jitter buffer
For each of the tests, both JB_TYPE_VOICE and JB_TYPE_CONTROL permutations
exist.
Review: https://reviewboard.asterisk.org/r/1815
(issue: ASTERISK-18964)
Reported by: Kris Shaw
Tested by: Kris Shaw, Matt Jordan
Matthew Jordan [Wed, 14 Mar 2012 17:48:40 +0000 (17:48 +0000)]
Fix incorrect jitter buffer overflow due to missed resynchronizations
When a change in time occurs, such that the timestamps associated with frames
being placed into an adaptive jitter buffer (implemented in jitterbuf.c)
are significantly different then the previously inserted frames, the jitter
buffer checks to see if it needs to be resynched to the new time frame. If
three consecutive packets break the threshold, the jitter buffer resynchs
itself to the new timestamps. This currently only occurs when history is
calculated, and hence only on JB_TYPE_VOICE frames.
JB_TYPE_CONTROL frames, on the other hand, are never passed to the history
calculations. Because of this, if the jump in time is greater then the
maximum allowed length of the jitter buffer, the JB_TYPE_CONTROL frames are
dropped and no resynchronization occurs. Alterntively, if the overfill
logic is not triggered, the JB_TYPE_CONTROL frame will be placed into the
buffer, but with a time reference that is not applicable. Subsequent
JB_TYPE_VOICE frames will quickly trigger the overflow logic until reads
from the jitter buffer reach the errant JB_TYPE_CONTROL frame.
This patch allows JB_TYPE_CONTROL frames to resynch the jitter buffer. As
JB_TYPE_CONTROL frames are unlikely to occur in multiples, it perform the
resynchronization on any JB_TYPE_CONTROL frame that breaks the resynch
threshold.
Note that this only impacts chan_iax2, as other consumers of the adaptive
jitter buffer use the abstract jitter buffer API, which does not use
JB_TYPE_CONTROL frames.
Review: https://reviewboard.asterisk.org/r/1814/
(closes issue ASTERISK-18964)
Reported by: Kris Shaw
Tested by: Kris Shaw, Matt Jordan
Patches:
jitterbuffer-2012-2-26.diff uploaded by Kris Shaw (license 5722)
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Richard Mudgett [Wed, 14 Mar 2012 17:39:45 +0000 (17:39 +0000)]
Fix Dial m and r options and forked calls generating warnings for voice frames.
When connected line support was added, the wait_for_answer() variable
single changed its meaning slightly. Unfortunately, the places where
single was used did not necessarily get updated to reflect that change.
Also audio/video frames were sent to all forked calls when the endpoints
were never made compatible.
* Don't pass audio/video media frames when the channels have not been made
compatible.
* Added handling of AST_CONTROL_SRCCHANGE to app_dial.c.
* Fixed app_dial.c passing on AST_CONTROL_HOLD because that frame can also
pass a requested MOH class.
(closes issue ASTERISK-16901)
Reported by: Chris Gentle
Matthew Jordan [Wed, 14 Mar 2012 14:40:44 +0000 (14:40 +0000)]
Force non-inlining of ao2_iterator_destroy when TEST_FRAMEWORK is enabled
In r357272, astobj2 was changed to automatically enable REF_DEBUG when the
TEST_FRAMEWORK flag was enabled. Unfortunately, some compilers (gcc 4.5.1
at least) will attempt to inline ao2_iterator_destroy in handle_astobj2_test.
This by itself is not a problem; unfortunately, the compiler believes that
there is a code path wherein an object allocated on the stack will be
free'd. As warnings are treated as errors, this prevents compilation of
astobj2.
This patch works around that by adding the noinline attribue to
ao2_iterator_destroy, but only if the TEST_FRAMEWORK flag is enabled.
Preventing inlining is only needed for the test method defined in astobj2,
which is also only enabled if TEST_FRAMEWORK is enabled.
Russell Bryant [Wed, 14 Mar 2012 10:56:53 +0000 (10:56 +0000)]
Fix bogus reads/writes of console log levels in asterisk.c
This patch updates the NUMLOGLEVELS define in logger.h to 32, to match the fact
that logger.c implements 32 log levels (because of the custom log level stuff).
asterisk.c uses this define to size an array of levels per remote console.
This array is modified in ast_console_toggle_loglevel(), which is called by the
"logger set level" CLI command. While the documentation for the CLI command
doesn't make it terribly obvious, you can use this CLI command to toggle a
custom log level on a remote console, as well. However, doing so led to an
invalid array index in asterisk.c.
This array is read from any time a log message is written to a console. So,
all custom log level messages resulted in a bogus read if a remote console
was connected.
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Russell Bryant [Wed, 14 Mar 2012 10:05:07 +0000 (10:05 +0000)]
Fix invalid reads/writes due to incorrect sizeof().
These few places in the code used sizeof() on h_addr in struct hostent.
This is sizeof(char *). The correct way to get the size of this address is to
use h_length. This error would result in reads/writes of 8 bytes instead of 4
on 64-bit machines.
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Russell Bryant [Wed, 14 Mar 2012 00:22:10 +0000 (00:22 +0000)]
Fix incorrect sizeof() usage in features.c.
This didn't actually result in a bug anywhere, luckily. The only place
where the result of these memcpys was used is in app_dial, and the only
field that it read out of ast_call_feature was the first one, which is an
int, so these memcpys always copied just enough to avoid a problem.
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Terry Wilson [Tue, 13 Mar 2012 20:43:19 +0000 (20:43 +0000)]
Fix setting CDR variables in the hangup extension
A previous CDR fix for setting CDR variables during a bridge via
custom dialplan features broke setting CDR variables in the
hangup extension. This patch fixes the issue.
Terry Wilson [Tue, 13 Mar 2012 20:06:57 +0000 (20:06 +0000)]
Make hints for invalid SIP devices return Unavail, not idle
This patch drastically simplifies the device state aggegation code.
The old method was not only overly complex, but also made it impossible
to return AST_DEVICE_INVALID from the aggregation code. The unit test
update is as a result of fixing that bug.
The SIP change stems from a bug introduced by removing a DNS lookup
for hostname-based SIP channels.
Tilghman Lesher [Tue, 13 Mar 2012 08:06:20 +0000 (08:06 +0000)]
Enable macros in 1.8 to find the next highest "h" extension in a context, like in 1.4.
This change restores functionality that was present in 1.4, when AEL macros
were implemented with the Macro dialplan application. Macros are fraught with
functionality issues, because they consume a large portion of the underlying
application stack. This limits the ability of AEL users to call many layers
of subroutines, an issue which Gosub does not have (originally tested to
100,000 levels deep). Therefore, starting in 1.6.0, AEL macros were
implemented with Gosub.
However, there were some implicit behaviors of Macro, which were not replicated
at the same time as with the transition to Gosub, one of which is documented in
the related issue. In particular, the "h" extension is designed to execute not
in the Macro context, but in the topmost calling context. Due to legacy issues
with a misapplied bugfix many years ago, when a macro exited in 1.4, it looks
in all calling contexts, bubbling up from the deepest level until it finds an
"h" extension.
Since AEL hides the complexity of the underlying dialplan logic from the AEL
programmer, it's reasonable to assume that this behavior should not change in
the transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we break
working AEL configurations in the transition to Asterisk 1.8 LTS. This fix
is the result, which implements a search for the "h" extension in all calling
Gosub contexts.
Fixes ASTERISK-19336
Patch: 20120308__ael_bugfix_for_trunk__2.diff (License #5003) by Tilghman Lesher
(with slight modifications for 1.8)
Massive changes in chan_unistim channel driver. Include many fixes in channel driver operation and add additional functionality:
* Added ability to use multiple lines on phone, so for one device in configuration multiple lines can be defined, it allows to have multiple calls on one phone, callwaiting and switching between calls.
* Added ability for translation on-screen menu to multiple languages. Tested on Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4, ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen menu of phone
* Other described in CHANGES file
Testing done by issue tracker users: ibercom, scsiborg, idarwin, TeknoJuce, c0rnoTa.
Tested on production system by Jonn Taylor (jonnt) using phone models: Nortel i2004, 1120E and 1140E.
Joshua Colp [Sat, 10 Mar 2012 20:06:46 +0000 (20:06 +0000)]
Transition app_page to using app_confbridge internally for the conference bridge portion of paging. This also adds a new 'announcement' option to ConfBridge user profiles.
Jonathan Rose [Thu, 8 Mar 2012 16:59:30 +0000 (16:59 +0000)]
Make transfer not ignore port information with SIP.
Attempting to transfer with SIP to an address like 1XXXXX@ip.ad.re.ss:5061 would fail
because port would be cut from the host string and ignored. This simply keeps chan_sip
from cutting off the port number during these kinds of transfers.
Sean Bright [Thu, 8 Mar 2012 16:21:49 +0000 (16:21 +0000)]
Add --enable-dev-mode=strict to configure.
Passing -Wshadow to gcc enables shadow warnings. From the gcc manual:
Warn whenever a local variable or type declaration shadows another
variable, parameter, type, or class member (in C++), or whenever a
built-in function is shadowed.
Asterisk will not currently compile with this option set, but a number of bugs
have been discovered by enabling this flag on specific files. The long-term
goal is to eliminate all of the suspect code that causes this warning to be
emitted.
Terry Wilson [Wed, 7 Mar 2012 15:08:08 +0000 (15:08 +0000)]
Add detection for ODBC WCHAR fields
Without detecting these types, cel_odbc blows up when the character
set for the table is utf8. This also wraps cdr_adaptive_odbc's use of
those types in the HAVE_ODBC_WCHAR #ifdef seen in other parts of the
code.
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Richard Mudgett [Tue, 6 Mar 2012 01:56:10 +0000 (01:56 +0000)]
Add dialtone_detect option for analog incoming calls.
For analog lines, enables Asterisk to use dialtone detection per channel
if an incoming call was hung up before it was answered. If dialtone is
detected, the call is hung up.
no: Disabled. (Default)
yes: Look for dialtone for 10000 ms after answer.
<number>: Look for dialtone for the specified number of ms after answer.
always: Look for dialtone for the entire call. Dialtone may return
if the far end hangs up first.
Richard Mudgett [Mon, 5 Mar 2012 21:48:32 +0000 (21:48 +0000)]
Setup DSP when SS7 call is connected or early media is available.
Outgoing SS7 calls fail to detect incoming DTMF so any bridged channel
that requires out-of-band DTMF will not work.
* Added sig_ss7_open_media() calls at appropriate places in sig_ss7.c.
The new call converts conditionaled out unconverted code and shows that
the code really did something useful.
* Improved some chan_dahdi DTMF debug messages to help track DTMF
handling.
(closes issue ASTERISK-19312)
Reported by: Igor Nikolaev
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Jonathan Rose [Mon, 5 Mar 2012 19:06:46 +0000 (19:06 +0000)]
Eliminate double close of file descriptor in manager.c
The process_output function in manager.c attempted to call fclose and close immediately
afterwards. Since fclose implies close, this resulted in a potential double free on file
descriptors. This patch changes that behavior and also adds error checking to fclose and
close depending on which was deemed necessary. Also error messages. Thanks to Rosen
Iliev for pointing out the location of the problem.
Kinsey Moore [Mon, 5 Mar 2012 16:00:32 +0000 (16:00 +0000)]
Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors
Asterisk was not setting pendinginvite in the upper half of
handle_request_invite such that the 4xx was retransmitted repeatedly even
though an ack was received for every retransmission.
(closes issue ASTERISK-19303)
Reported by: Jon Tsiros
Patches:
fix-19303.patch uploaded by Jeremiah Gowdy (license 6358)
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Terry Wilson [Fri, 2 Mar 2012 23:29:53 +0000 (23:29 +0000)]
Fix unused-but-set-variable warnings
All of these were pretty obviously unused. Some were unused because
the code that used them was #if 0'd. In those cases, I just commented
out the unused-but-set variables.
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Kinsey Moore [Fri, 2 Mar 2012 21:06:12 +0000 (21:06 +0000)]
Fix case-sensitivity for device-specific event subscriptions and CCSS
This change fixes case-sensitivity for device-specific subscriptions such that
the technology identifier is case-insensitive while the remainder of the device
string is still case-sensitive. This should also preserve the original case of
the device string as passed in to the event system. CCSS is the only feature
affected as it is the only consumer of device-specific event subscriptions.
The second part of this patch addresses similar case-sensitivity issues within
CCSS itself that prevented it from functioning correctly after the fix to the
events system.
This adds a unit test to verify that the event system works as expected.
Richard Mudgett [Fri, 2 Mar 2012 18:38:49 +0000 (18:38 +0000)]
Remove ISDN hold restriction for non-bridged calls.
The check if an ISDN call is bridged before it could be placed on hold is
not necessary and is overly restrictive. The check was originally done to
prevent problems with call transfers in case a user tried to transfer a
call connected to an application to another call connected to an
application. The ISDN transfer code has not required this restriction for
quite some time because ECT could transfer any two active calls to each
other.
* Remove ISDN hold restriction for calls connected to applications.
* Made ast_waitfordigit_full() ignore AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD instead of generating a warning message.
Jonathan Rose [Fri, 2 Mar 2012 16:57:12 +0000 (16:57 +0000)]
Adds a transfer callee on hangup option (like with Dial option F) to queues.
This should (and does in my testing) act just like the Dial option of the same name.
This allows a queue member to be transfered to the next priority (no args), or to
a context/extension/priority similar to goto (with args context^extension^priority)
when a caller hangs up on them.
(closes issue ASTERISK-19283)
Reported by: To
Patches:
queue_f-v3.diff uploaded by To (license 6347)
Review: https://reviewboard.asterisk.org/r/1785/
Mark Michelson [Fri, 2 Mar 2012 01:33:06 +0000 (01:33 +0000)]
Fix race condition that can cause important control frames (such as a hangup) to be missed.
This takes two actions.
1. Move the reading of the alertpipe in __ast_read() to immediately before the
removal of frames from the readq. This means we won't do something silly like
read from the alertpipe, then ignore the fact that there's a frame to get from
the readq since channel's fdno is the AST_TIMING_FD.
2. When ast_settimeout() sets the rate to 0 and the timingfunc to NULL, if the
channel's fdno is the AST_TIMING_FD, then set the fdno to -1. This is because
if the rate is 0 and the timingfunc is NULL, it means that the channel's timing
fd is being invalidated, so any pending reads should not occur.
This may actually solve more issues than the referenced one below, but it's not
known at this time for sure.
(closes issue ASTERISK-19223)
reported by Frank-Michael Wittig
Kinsey Moore [Thu, 1 Mar 2012 14:22:01 +0000 (14:22 +0000)]
Prevent outbound SIP NOTIFY packets from displaying a port of 0
In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which changed the
behavior of ast_find_ourip such that port number was wiped out. This caused
the port in internip (which is used for Contact and Call-ID on NOTIFYs) to be
0. This change causes ast_find_ourip to be port-preserving again.
(closes issue ASTERISK-19430)
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Walter Doekes [Wed, 29 Feb 2012 20:41:38 +0000 (20:41 +0000)]
Update stringfield documentation for removed second va_list in favor of va_copy.
In r320946, the second va_list that was passed to ast_string_field_build_va
and friends, was removed. This patch updates the documentation to reflect that.
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Walter Doekes [Wed, 29 Feb 2012 19:48:33 +0000 (19:48 +0000)]
Fix copying of CDR(accountcode) to local channels.
In r203638, during the addition of the Channel Event Logging, in mid-2009, this
got broken in trunk and ended up in asterisk 1.8 and higher. This fixes so the
CDR(accountcode) from the calling channel is available to dialed channels again
as well as showing up properly in the CDR's.
(closes issue ASTERISK-19384)
Reported by: jamicque
Patches: accountcode.patch (License #6033) by jamicque
Review: https://reviewboard.asterisk.org/r/1775/
Reviewed by: Richard Mudgett
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Jonathan Rose [Tue, 28 Feb 2012 22:31:24 +0000 (22:31 +0000)]
Adding transport=udp to sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE
(issue ASTERISK-19352)
Reported by: jamicque
Patches:
asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026)
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Jonathan Rose [Tue, 28 Feb 2012 22:01:45 +0000 (22:01 +0000)]
Add additional character type types to supported data types for cdr_adaptive_odbc
The reporter was uable to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so
this patch adds those along with some other character types to the list of types
cdr_adaptive_odbc will work using the varchar conditions. The problem wasn't really
UTF8 characters as much as it was a failure to respond to the exact type that was
declared/in use on that database.
(closes issue ASTERISK-19334)
Reported By: Igor Nikolaev
Patches:
cdr_adaptive_odbc.patch uploaded by Igor Nikolaev (license 6236)
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Tilghman Lesher [Tue, 28 Feb 2012 21:26:34 +0000 (21:26 +0000)]
Correctly reset the dialplan priority.
When the stack frame is allocated, we save the address to which we should
return, when the Gosub returns. However, if we just want to restore the
priority, then we need to subtract 1 before setting it. Otherwise, when
a Gosub goes to a nonexistent address, it will skip a priority in the
dialplan. This is because when we return from an application, the PBX
increments the priority for us.
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Jonathan Rose [Tue, 28 Feb 2012 18:23:02 +0000 (18:23 +0000)]
Changes transport option in sip.conf so that using multiple instances doesn't stack.
Prior to this patch, Using "transport=" multiple times would cause them to add to one
another like allow/deny. This patch changes that behavior to simply use the transport
option specified last. Also, if no transport option is applied now, the default will
automatically be UDP.
(closes ASTERISK-19352)
Reported by: jamicque
Patches:
asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026)
issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes (license 5674)
Review: https://reviewboard.asterisk.org/r/1745/diff/#index_header
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The OBJ_NOLOCK and AO2_ITERATOR_DONTLOCK flags have a slight meaning
change. They no longer mean that the object is protected by an external
mechanism. They mean the lock associated with the object has already been
manually obtained by one of the ao2_lock calls. This change is necessary
for RWLOCK support since they are not reentrant. Also an operation on an
ao2 container may require promoting a read lock to a write lock by
releasing the already held read lock to re-acquire as a write lock.
API calls to be more flexible and to allow an anticipated enhancement to
control linking duplicate objects into a container.
The changes to format.c and format_cap.c are taking advantange of the new
ao2 locking options to simplify the use of the format capabilities
containers.
Kevin P. Fleming [Tue, 28 Feb 2012 14:47:23 +0000 (14:47 +0000)]
Make COMPILE_DOUBLE magic actually work.
The build system has some special magic to ensure that if Asterisk is built
with --enable-dev-mode *and* DONT_OPTIMIZE, that all the source is still compiled
with the optimizer enabled (even though the result will be thrown away), because
the compiler is able to find a great deal of coding errors and bugs as a result
of running its optimizers. Unfortunately at some point this mode got broken,
and the 'throwaway' compile of the code was no longer done with the optimizer
enabled. This patch corrects that problem.
........
Merged revisions 357212 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 357213 from http://svn.asterisk.org/svn/asterisk/branches/10
Richard Mudgett [Tue, 28 Feb 2012 00:42:38 +0000 (00:42 +0000)]
Add ability to clone ao2 containers.
Occasionally there is a need to put all objects in one container also into
another container.
Some reasons you might need to do this:
1) You need to reconfigure a container. You would do this by creating a
new container with the new configuration and ao2_container_dup the old
container into it. Then replace the old container with the new. Then
destroy the old container.
2) You need the contents of a container to remain stable while operating
on all of the objects. You would do this by creating a cloned container
of the original with ao2_container_clone. The cloned container is a
snapshot of the objects at the time of the cloning. When done, just
destroy the cloned container.
Richard Mudgett [Tue, 28 Feb 2012 00:17:19 +0000 (00:17 +0000)]
Fix ast_channel allocation init setting priority to -1 instead of 1.
* Fix opaquification conversion error.
(closes issue ASTERISK-19424)
Reported by: Jeremy Pepper
Patches:
asterisk-19424-initialize_priority_regression.diff (license #5026) patch uploaded by Michael L. Young
projects / thirdparty / asterisk.git / log
