[asterisk-commits] qwell: branch 10-digiumphones r368781 - in /branches/10-digiumphones: ./ addo...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Jun 11 16:54:10 CDT 2012
Author: qwell
Date: Mon Jun 11 16:53:41 2012
New Revision: 368781
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=368781
Log:
Multiple 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
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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.
(closes issue ASTERISK-18243)
Reported by: mjordan
Tested by: mjordan
Review: https://reviewboard.asterisk.org/r/1892
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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.
(closes issue ASTERISK-19643)
reported by: Shaun Ruffell
Patches:
0001-codec_dahdi-Block-on-frameout-the-hardware-has-enoug.patch
uploaded by Shaun Ruffell (license 5417)
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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
Review: https://reviewboard.asterisk.org/r/1911
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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.
(issue AST-876)
Review: https://reviewboard.asterisk.org/r/1899/
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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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(issue AST-876)
Merged revisions 366791 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r366881 | mjordan | 2012-05-18 09:01:56 -0500 (Fri, 18 May 2012) | 65 lines
Fix a variety of memory leaks
This patch addresses a number of memory leaks in a variety of modules that were
found by a static analysis tool. A brief summary of the changes:
* app_minivm: free ast_str objects on off nominal paths
* app_page: free the ast_dial object if the requested channel technology
cannot be appended to the dialing structure
* app_queue: if a penalty rule failed to match any existing rule list
names, the created rule would not be inserted and its memory
would be leaked
* app_read: dispose of the created silence detector in the presence of
off nominal circumstances
* app_voicemail: dispose of an allocated unique ID field for MWI event
un-subscribe requests in off nominal paths; dispose of
configuration objects when using the secret.conf option
* chan_dahdi: dispose of the allocated frame produced by ast_dsp_process
* chan_iax2: properly unref peer in CLI command "iax2 unregister"
* chan_sip: dispose of the allocated frame produced by sip_rtp_read's
call of ast_dsp_process; free memory in parse unit tests
* func_dialgroup: properly deref ao2 object grhead in nominal path of
dialgroup_read
* func_odbc: free resultset in off nominal paths of odbc_read
* cli: free match_list in off nominal paths of CLI match completion
* config: free comment_buffer/list_buffer when configuration file load
is unchanged; free the same buffers any time they were
created and config files were processed
* data: free XML nodes in various places
* enum: free context buffer in off nominal paths
* features: free ast_call_feature in off nominal paths of applicationmap
config processing
* netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct
that is allocated by the method. Failures in
ast_sockaddr_resolve could result in the users of the method
not knowing whether or not the buffer was allocated. The
method will now not allocate the ast_sockaddr struct if it
will return failure.
* pbx: cleanup hash table traversals in off nominal paths; free
ignore pattern buffer if it already exists for the specified
context
* xmldoc: cleanup various nodes when we no longer need them
* main/editline: various cleanup of pointers not being freed before being
assigned to other memory, cleanup along off nominal paths
* menuselect/mxml: cleanup of value buffer for an attribute when that attribute
did not specify a value
* res_calendar*: responses are allocated via the various *_request method
returns and should not be allocated in the various
write_event methods; ensure attendee buffer is freed if no
data exists in the parsed node; ensure that calendar objects
are de-ref'd appropriately
* res_jabber: free buffer in off nominal path
* res_musiconhold: close the DIR* object in off nominal paths
* res_rtp_asterisk: if we run out of ports, close the rtp socket object and free
the rtp object
* res_srtp: if we fail to create the session in libsrtp, destroy the
temporary ast_srtp object
(issue ASTERISK-19665)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1922
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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
(issue ASTERISK-19665)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1922/
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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.
Review: https://reviewboard.asterisk.org/r/1900/
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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
Review: https://reviewboard.asterisk.org/r/1939
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Merged revisions 367362 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r367417 | mmichelson | 2012-05-23 15:29:03 -0500 (Wed, 23 May 2012) | 7 lines
Only call SSL_CTX_free if DO_SSL is defined.
Thanks to Paul Belanger for pointing out this error.
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Merged revisions 367416 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r367470 | rmudgett | 2012-05-23 18:16:49 -0500 (Wed, 23 May 2012) | 9 lines
Fix WaitExten(x,m(musicclass)) string termination.
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.
(closes issue ASTERISK-19899)
Reported by: Florian Gilcher
Tested by: Florian Gilcher
patches:
asterisk-19899.diff uploaded by mjordan (license 6283)
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r367679 | rmudgett | 2012-05-24 17:29:23 -0500 (Thu, 24 May 2012) | 34 lines
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.
(closes issue ASTERISK-19511)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1920/
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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)
Review: https://reviewboard.asterisk.org/r/1923/
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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)
Review: https://reviewboard.asterisk.org/r/1949/
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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.
Review: https://reviewboard.asterisk.org/r/1811/
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Merged revisions 368218 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r368310 | rmudgett | 2012-06-01 18:24:25 -0500 (Fri, 01 Jun 2012) | 15 lines
Fix deadlock when Gosub used with alternate dialplan switches.
Attempting to remove a channel from autoservice with the channel lock held
will result in deadlock.
* Restructured gosub_exec() to not call ast_parseable_goto() and
ast_exists_extension() with the channel lock held.
(closes issue ASTERISK-19764)
Reported by: rmudgett
Tested by: rmudgett
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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.
(closes issue ASTERISK-19537)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1915/
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Merged revisions 368405 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r368470 | rmudgett | 2012-06-04 16:11:42 -0500 (Mon, 04 Jun 2012) | 10 lines
Document BLINDTRANSFER behavior change.
(issue ASTERISK-19322)
(closes issue ASTERISK-19875)
Reported by: call
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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.
(closes issue ABE-2862)
Reported by: Vlad Povorozniuc
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Merged revisions 368567 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r368587 | kmoore | 2012-06-06 11:09:10 -0500 (Wed, 06 Jun 2012) | 12 lines
Ensure overlapping hold flags do not conflict
When changing between different modes of hold, the flags were not being
cleared out properly causing a failure to change hold states.
(closes issue ASTERISK-19919)
Patch-by: Morten Tryfoss
Reported-by: Morten Tryfoss
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Merged revisions 368586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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