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.
........
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.
........
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
........
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.
........
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.
........
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.
........
Merged revisions 360309 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Wed, 21 Mar 2012 13:28:17 +0000 (13:28 +0000)]
Ensure Asterisk sends a BYE when pending on the final response to a re-INVITE
When Asterisk detects a hangup and cannot send a BYE due to a pending
INVITE, it sets the pendingbye flag and waits for the final response to that
INVITE. When the response is received, it transmits the BYE. If, however,
that INVITE request is a pending re-INVITE, it needs to first send a CANCEL
request to terminate the pending re-INVITE. In that circumstance, Asterisk
was, in some scenarios, clearing the pendingbye flag after processing the
CANCEL request and not checking for a pending BYE when receiving the final
487 response to the INVITE.
This patch ensures that if the pendingbye flag is set, it is honored
regardless of the nature of the INVITE request currently in flight.
(closes issue ASTERISK-19365)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Patches:
bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license 6283)
Kinsey Moore [Tue, 20 Mar 2012 20:37:46 +0000 (20:37 +0000)]
Prevent Echo() from relaying control, null, and modem frames
Echo()'s description states that it echoes audio, video, and DTMF except for #
while it actually echoes any frame that it receives other than DTMF #. This
was causing frame storms in the test suite in some circumstances where Echo()
was attached to both ends of a pair of local channels and control frames
were being periodically generated. Echo()'s behavior and description have
been modifed so that it only echoes media and non-# DTMF frames.
........
Merged revisions 360033 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Sean Bright [Tue, 20 Mar 2012 18:11:41 +0000 (18:11 +0000)]
chan_iax2: Emit Port alongside Post in PeerStatus AMI Event.
The PeerStatus event for IAX2 channels currently includes a header named Post
which should have been Port. So include Port along with Post when emitting the
event. We'll remove Post in trunk.
Richard Mudgett [Tue, 20 Mar 2012 17:25:44 +0000 (17:25 +0000)]
Allow AMI action callback to be reentrant.
Fix AMI module reload deadlock regression from ASTERISK-18479 when it
tried to fix the race between calling an AMI action callback and
unregistering that action. Refixes ASTERISK-13784 broken by
ASTERISK-17785 change.
Locking the ao2 object guaranteed that there were no active callbacks that
mattered when ast_manager_unregister() was called. Unfortunately, this
causes the deadlock situation. The patch stops locking the ao2 object to
allow multiple threads to invoke the callback re-entrantly. There is no
way to guarantee a module unload will not crash because of an active
callback. The code attempts to minimize the chance with the registered
flag and the maximum 5 second delay before ast_manager_unregister()
returns.
The trunk version of the patch changes the API to fix the race condition
correctly to prevent the module code from unloading from memory while an
action callback is active.
* Don't hold the lock while calling the AMI action callback.
(closes issue ASTERISK-19487)
Reported by: Philippe Lindheimer
Jonathan Rose [Fri, 16 Mar 2012 20:20:25 +0000 (20:20 +0000)]
Prevent chanspy from binding to zombie channels
This patch addresses a bug with chanspy on local channels which roughly 50% of the time
would create a situation where chanspy can latch onto a zombie channel, keeping the zombie
alive forever and causing the channel doing the spying to never be able to hang up.
Matthew Jordan [Thu, 15 Mar 2012 19:06:09 +0000 (19:06 +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
........
Merged revisions 359706 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Thu, 15 Mar 2012 18:50:17 +0000 (18:50 +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
........
Merged revisions 359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
........
Merged revisions 359656 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Thu, 15 Mar 2012 18:22:01 +0000 (18:22 +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 23:28:32 +0000 (23:28 +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.
........
Merged revisions 359486 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Russell Bryant [Wed, 14 Mar 2012 23:02:43 +0000 (23:02 +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).
........
Merged revisions 359457 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Russell Bryant [Wed, 14 Mar 2012 22:37:01 +0000 (22:37 +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.
........
Merged revisions 359452 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Wed, 14 Mar 2012 22:28:35 +0000 (22:28 +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.
........
Merged revisions 359451 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Wed, 14 Mar 2012 17:42:16 +0000 (17:42 +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)
........
Merged revisions 359356 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Wed, 14 Mar 2012 17:24:00 +0000 (17:24 +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
Russell Bryant [Wed, 14 Mar 2012 10:54:50 +0000 (10:54 +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.
........
Merged revisions 359259 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Russell Bryant [Wed, 14 Mar 2012 10:04:03 +0000 (10:04 +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.
........
Merged revisions 359211 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Russell Bryant [Wed, 14 Mar 2012 00:21:18 +0000 (00:21 +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.
........
Merged revisions 359069 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Terry Wilson [Tue, 13 Mar 2012 20:36:06 +0000 (20:36 +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 20:00:03 +0000 (20:00 +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 07:48:01 +0000 (07:48 +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)
Jonathan Rose [Thu, 8 Mar 2012 16:50:45 +0000 (16:50 +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.
Terry Wilson [Wed, 7 Mar 2012 15:07:04 +0000 (15:07 +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.
........
Merged revisions 358435 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Mon, 5 Mar 2012 21:38:50 +0000 (21:38 +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
........
Merged revisions 358260 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Mon, 5 Mar 2012 18:58:40 +0000 (18:58 +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 15:59:46 +0000 (15:59 +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)
........
Merged revisions 358115 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Terry Wilson [Fri, 2 Mar 2012 23:28:21 +0000 (23:28 +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.
........
Merged revisions 358029 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Fri, 2 Mar 2012 21:03:11 +0000 (21:03 +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 18:37:15 +0000 (18:37 +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.
Mark Michelson [Fri, 2 Mar 2012 01:05:23 +0000 (01:05 +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 14:18:49 +0000 (14:18 +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)
........
Merged revisions 357665 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Walter Doekes [Wed, 29 Feb 2012 20:39:39 +0000 (20:39 +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.
Walter Doekes [Wed, 29 Feb 2012 19:43:02 +0000 (19:43 +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
........
Merged revisions 357575 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Tue, 28 Feb 2012 22:29:47 +0000 (22:29 +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)
........
Merged revisions 357490 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Tue, 28 Feb 2012 21:52:06 +0000 (21:52 +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)
........
Merged revisions 357455 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Tilghman Lesher [Tue, 28 Feb 2012 21:21:14 +0000 (21:21 +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.
........
Merged revisions 357416 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Tue, 28 Feb 2012 18:11:15 +0000 (18:11 +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
........
Merged revisions 357266 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kevin P. Fleming [Tue, 28 Feb 2012 14:46:15 +0000 (14:46 +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
Matthew Jordan [Sat, 25 Feb 2012 17:22:14 +0000 (17:22 +0000)]
Fix crash in app_voicemail during close_mailbox
In r354890, a memory leak in app_voicemail was fixed by properly disposing of
the allocated heard/deleted pointers. However, there are situations,
particularly when no messages are found in a folder, where these pointers are
not allocated and not NULL. In that case, an invalid free would be attempted,
which could crash app_voicemail. As there are a number of code paths where
this could occur, this patch uses the number of messages detected in the folder
before it attempts to free the pointers. This resolves the crash detected in
the Asterisk Test Suite's check_voicemail_nominal test.
........
Merged revisions 356797 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Fri, 24 Feb 2012 17:42:53 +0000 (17:42 +0000)]
Remove srtp_shutdown from res_srtp
The patch for ASTERISK-19253 included properly shutting down the libsrtp
library in the case of module unload. Unfortunately, not all distributions
have the srtp_shutdown call. As such, this patch removes calling
srtp_shutdown.
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Merged revisions 356650 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Fri, 24 Feb 2012 15:07:41 +0000 (15:07 +0000)]
Allow SRTP policies to be reloaded
Currently, when using res_srtp, once the SRTP policy has been added to the
current session the policy is locked into place. Any attempt to replace an
existing policy, which would be needed if the remote endpoint negotiated a new
cryptographic key, is instead rejected in res_srtp. This happens in particular
in transfer scenarios, where the endpoint that Asterisk is communicating with
changes but uses the same RTP session.
This patch modifies res_srtp to allow remote and local policies to be reloaded
in the underlying SRTP library. From the perspective of users of the SRTP API,
the only change is that the adding of remote and local policies are now added
in a single method call, whereas they previously were added separately. This
was changed to account for the differences in handling remote and local
policies in libsrtp.
Review: https://reviewboard.asterisk.org/r/1741/
(closes issue ASTERISK-19253)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Patches:
srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283)
(with some small modifications for this check-in)
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Merged revisions 356604 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Thu, 23 Feb 2012 19:52:39 +0000 (19:52 +0000)]
Fix blind transfer parking issues if the dialed extension is not recognized as a parking extension.
Custom parking extensions may not be coded such that the first and only
extension priority is the Park application. These custom parking
extensions will not be recognized as parking extensions. When a call is
blind transferred to an extension that is not recognized as a parking
extension, the normal blind transfer code causes the transferred channel
to start executing dialplan. Calls that get parked in this manner do not
know the original channel name that parked the call so the original parker
could never be called back if the parked call is not retrieved before the
timeout time. The parking space is also announced to the call being
parked as a side effect of not knowing the original parking channel.
* Fix handling of BLINDTRANSFER channel variable for call parking.
* Fixed SIP blind transfer using the wrong dialplan context variable to
check for the parking extension.
Mark Michelson [Thu, 23 Feb 2012 15:40:23 +0000 (15:40 +0000)]
Fix ACK routing for non-2xx responses.
When we send an ACK for a 2xx response to an INVITE, we are supposed
to use the learned route set. However, when we receive a non-2xx final
response to an INVITE, we are supposed to send the ACK to the same place
we initially sent the INVITE.
We had been doing this up until the changes went in that would build a route
set from provisional responses. That introduced a regression where we would
use the learned route set under all circumstances.
With this change, we now will set the destination of our ACK based on the
invitestate. If it is INV_COMPLETED then that means that we have received
a non-2xx final response (INV_TERMINATED indicates a 2xx response was received).
If it is INV_CANCELLED, then that means the call is being canceled, which
means that we should be ACKing a 487 response.
The other change introduced here is setting the invitestate to INV_CONFIRMED
when we send an ACK *after* the reqprep instead of before. This way, we can
tell in reqprep more easily what the invitestate is prior to sending the ACK.
(closes issue ASTERISK-19389)
reported by Karsten Wemheuer
patches:
ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049)
(with some slight modifications prior to commit)
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Merged revisions 356475 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Terry Wilson [Wed, 22 Feb 2012 21:18:22 +0000 (21:18 +0000)]
Track module use count for res_calendar
If the res_calendar module was followed immediately by one of the
calendar tech modules and "core stop gracefully" was run, Asterisk
would crash.
This patch adds use count tracking for res_calendar so that it is
unloaded after the tech modules when shutting down gracefully. It
is now not possible to unload all the of the calendar modules via
"module unload res_calednar.so", but it is still possible to unload
them all via "module unload -h res_calendar.so".
Fix potential buffer overrun and memory leak when executing "sip show peers"
The "sip show peers" command uses a fix sized array to sort the current peers
in the peers ao2_container. The size of the array is based on the current
number of peers in the container. However, once the size of the array is
determined, the number of peers in the container can change, as the peers
container is not locked. This could cause a buffer overrun when populating
the array, if peers were added to the container after the array was created.
Additionally, a memory leak of the allocated array would occur if a user
caused the _show_peers method to return CLI_SHOWUSAGE.
We now create a snapshot of the current peers using an ao2_callback with the
OBJ_MULTIPLE flag. This size of the array is set to the number of peers
that the iterator will iterate over; hence, if peers are added or removed
from the peers container it will not affect the execution of the "sip show
peers" command.
Review: https://reviewboard.asterisk.org/r/1738/
(closes issue ASTERISK-19231)
(closes issue ASTERISK-19361)
Reported by: Thomas Arimont, Jamuel Starkey
Tested by: Thomas Arimont, Jamuel Starkey
Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan (license 6283)
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Sean Bright [Mon, 20 Feb 2012 18:39:22 +0000 (18:39 +0000)]
Remove spurious warning when 'qualifyfreqnotok' is set successfully.
(closes issue ASTERISK-17176)
Reported by: John Covert
Tested by: Sean Bright
Patches:
chan_iax2.c.qualifyfreqnotok.patch uploaded by John Covert (license 5512)
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Merged revisions 355997 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Alec L Davis [Sat, 18 Feb 2012 07:58:43 +0000 (07:58 +0000)]
push 'outgoing' flag from sig_XXX up to chan_dahdi
'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag.
sig_pri and sig_ss7 were keeping 'outgoing' flag insync.
Now provides a callback for all the low level sig_XXX modules.
(issue ASTERISK-19316)
alecdavis (license 585)
Reported by: Jeremy Pepper
Tested by: alecdavis
projects / thirdparty / asterisk.git / log
