Richard Mudgett [Wed, 6 Jun 2012 01:10:10 +0000 (01:10 +0000)]
Fix parked call performing a DTMF blind transfer after being retrieved.
When a parked call was retrieved from the parking lot, it could not do a
blind transfer because it caused the involved calls to be hung up
unconditionally.
* Made the ParkedCall application return the ast_bridge_call() return
value.
Kinsey Moore [Tue, 5 Jun 2012 15:19:58 +0000 (15:19 +0000)]
Ensure that pages and emails are sent using RFC822-compliant date format
When localization was added to app_voicemail, these headers were altered
when they should have remained in en_US format for RFC compliance. This
reverts the changes to those two lines.
(closes issue ASTERISK-19876)
........
Merged revisions 368520 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Mon, 4 Jun 2012 22:02:26 +0000 (22:02 +0000)]
Relay proper SIP responses on calling side.
Revision 351130 broke corect HANGUPCAUSE setting
for the 404 case in chan_sip. Other cases were also
potentially broken. This patch fixes the relaying
of causes to be what they used to be.
(closes issue ASTERISK-19914)
Reported by Pavel Troller
Tested by Walter Doekes (via a reviewboard test to be committed later)
Patches:
chan_sip.diff uploaded by Pavel Troller (license #6302)
........
Merged revisions 368498 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Mon, 4 Jun 2012 19:08:52 +0000 (19:08 +0000)]
Fix potential deadlock between masquerade and chan_local.
* Restructure ast_do_masquerade() to not hold channel locks while it calls
ast_indicate().
* Simplify many calls to ast_do_masquerade() since it will never return a
failure now. If it does fail internally because a channel driver callback
operation failed, the only thing ast_do_masquerade() can do is generate a
warning message about strange things may happen and press on.
* Fixed the call to ast_bridged_channel() in ast_do_masquerade(). This
change fixes half of the deadlock reported in ASTERISK-19801 between
masquerades and chan_iax.
* 'Unsupported media type' is only reported when that is in fact the case,
not when a supported media type is included in an 'm' line that has an
invalid format.
* All warning messages related to parsing 'm' lines now include the 'm' line contents.
* (minor bugfix) newline added to port-number-zero warning messages.
* Warning messages improved to use RFC-specified terminology for various items.
* Warnings for offers that include more than one port for a single media type now
include the media type.
Add documentation to function CHANNEL for options echocan_mode and buffers
The ability to set "echocan_mode" and "buffers" through the dialplan was added
to chan_dahdi some time ago. This patch adds some documentation to
func_channel.
(Closes issue ASTERISK-19911)
Reported by: Dale Noll
Tested by: Michael L. Young
Patches:
asterisk-19911-branch18.diff uploaded by Michael L. Young (license 5026)
Richard Mudgett [Wed, 30 May 2012 17:39:24 +0000 (17:39 +0000)]
Fix deadlock when executing CLI "pri show channels" and "ss7 show channels" commands.
* Fix sig_pri_lock_owner() to avoid deadlock properly.
* Code pri_grab() better.
* Fix sig_ss7_lock_owner() to avoid deadlock properly.
* Code ss7_grab() better.
(closes issue ASTERISK-19854)
Reported by: Jaxon
Patches:
jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded by rmudgett (Modified to do the same thing to sig_ss7)
Tested by: Jaxon
........
Merged revisions 367976 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Tue, 29 May 2012 22:28:55 +0000 (22:28 +0000)]
Coverity Report: Fix issues for error type REVERSE_INULL (deprecated modules)
* Fix only issue pointed out by deprecated_REVERSE_INULL.txt for
app_meetme.c in find_user().
* Change use of %i to %d in sscanf() in find_user(). The use of %i gives
unexpected parsing because it can accept hex, octal, and decimal integer
formats.
* Changed other uses of %i in app_meetme() to use %d for consistency.
(issue ASTERISK-19648)
Reported by: Matt Jordan
........
Merged revisions 367906 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Tue, 29 May 2012 18:33:20 +0000 (18:33 +0000)]
AST-2012-008: Fix remote crash vulnerability in chan_skinny
When a skinny session is unregistered, the corresponding device pointer is set
to NULL in the channel private data. If the client was not in the on-hook state
at the time the connection was closed, the device pointer can later be
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)
Michael L. Young [Fri, 25 May 2012 02:29:26 +0000 (02:29 +0000)]
Fix pvt_sip for inbound call to use peer's allowtransfer setting
The pvt_sip allowtransfer was not being set to that of the peer's setting.
Therefore, the global allowtransfer setting was being used instead which would
lead to calls not being transfered if the global setting was set to 'no' despite
the setting on the peer being 'yes' and vice versa, calls would be allowed to
transfer even if the peer's setting was 'no' but the global setting was 'yes'.
(Closes issue ASTERISK-19856)
Reported by: Jacek
Tested by: Michael L. Young, Jacek
Patches:
issue-asterisk-19856-branch10-v3.diff uploaded by
Michael L. Young (license 5026)
Richard Mudgett [Thu, 24 May 2012 22:29:23 +0000 (22:29 +0000)]
Fix Dial I option ignored if dial forked and one fork redirects.
The Dial and Queue I option is intended to block connected line updates
and redirecting updates. However, it is a feature that when a call is
locally redirected, the I option is disabled if the redirected call runs
as a local channel so the administrator can have an opportunity to setup
new connected line information. Unfortunately, the Dial and Queue I
option is disabled for *all* forked calls if one of those calls is
redirected.
* Make the Dial and Queue I option apply to each outgoing call leg
independently. Now if one outgoing call leg is locally redirected, the
other outgoing calls are not affected.
* Made Dial not pass any redirecting updates when forking calls.
Redirecting updates do not make sense for this scenario.
* Made Queue not pass any redirecting updates when using the ringall
strategy. Redirecting updates do not make sense for this scenario.
* Fixed deadlock potential with chan_local when Dial and Queue send
redirecting updates for a local redirect.
* Converted the Queue stillgoing flag to a boolean bitfield.
Matthew Jordan [Thu, 24 May 2012 13:32:33 +0000 (13:32 +0000)]
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.
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.
........
Merged revisions 367469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Wed, 23 May 2012 13:25:04 +0000 (13:25 +0000)]
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
Terry Wilson [Tue, 22 May 2012 17:21:51 +0000 (17:21 +0000)]
Fix race condition for CEL LINKEDID_END event
This patch fixes to situations that could cause the CEL LINKEDID_END event to
be missed.
1) During a core stop gracefully, modules are unloaded when ast_active_channels
== 0. The LINKDEDID_END event fires during the channel destructor. This means
that occasionally, the cel_* module will be unloaded before the channel is
destroyed. It seemed generally useful to wait until the refcount of all
channels == 0 before unloading, so I added a channel counter and used it in the
shutdown code.
2) During a masquerade, ast_channel_change_linkedid is called. It calls
ast_cel_check_retire_linkedid which unrefs the linkedid in the linkedids
container in cel.c. It didn't ref the new linkedid. Now it does.
Terry Wilson [Tue, 22 May 2012 16:17:46 +0000 (16:17 +0000)]
Resolve crash in subscribing for MWI notifications
ASTOBJ_UNREF sets the variable to NULL after unreffing it, so the variable
should definitely not be used after that. To solve this in the two cases
that affect subscribing for MWI notifications, we instead save the ref
locally, and unref them in the error conditions.
(closes issue ASTERISK-19827)
Reported by: B. R
Review: https://reviewboard.asterisk.org/r/1940/
........
Merged revisions 367266 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Fri, 18 May 2012 17:00:14 +0000 (17:00 +0000)]
Fix memory leak of SSL_CTX structures in TLS core.
SSL_CTX structures were allocated but never freed. This was a bigger
issue for clients than servers since new SSL_CTX structures could be
allocated for each connection. Servers, on the other hand, typically
set up a single SSL_CTX for their lifetime.
This is solved in two ways:
1. In __ssl_setup(), if a tcptls_cfg has an ssl_ctx on it, it is
freed so that a new one can take its place.
2. A companion to ast_ssl_setup() called ast_ssl_teardown() has
been added so that servers can properly free their SSL_CTXs.
(issue ASTERISK-19278)
........
Merged revisions 367002 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Fri, 18 May 2012 15:45:42 +0000 (15:45 +0000)]
Fix more memory leaks
This patch adds to what was fixed in r366880. Specifically, it addresses the
following:
* chan_sip: dispose of an allocated frame in off nominal code paths in
sip_rtp_read
* func_odbc: when disposing of an allocated resultset, ensure that any rows
that were appended to that resultset are also disposed of
* cli: free the created return string buffer in another off nominal code
path
Kinsey Moore [Fri, 18 May 2012 14:18:47 +0000 (14:18 +0000)]
Reorder and renumber tests appropriately
It appears that a patch did not apply properly when adding tests 12 and
13 and test 11 was duplicated. These tests have been reordered and
renumbered such that they make sense.
........
Merged revisions 366882 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Fri, 18 May 2012 14:01:56 +0000 (14:01 +0000)]
Fix a variety of memory leaks
This patch addresses a number of memory leaks in a variety of modules that were
found by a static analysis tool. A brief summary of the changes:
* app_minivm: free ast_str objects on off nominal paths
* app_page: free the ast_dial object if the requested channel technology
cannot be appended to the dialing structure
* app_queue: if a penalty rule failed to match any existing rule list
names, the created rule would not be inserted and its memory
would be leaked
* app_read: dispose of the created silence detector in the presence of
off nominal circumstances
* app_voicemail: dispose of an allocated unique ID field for MWI event
un-subscribe requests in off nominal paths; dispose of
configuration objects when using the secret.conf option
* chan_dahdi: dispose of the allocated frame produced by ast_dsp_process
* chan_iax2: properly unref peer in CLI command "iax2 unregister"
* chan_sip: dispose of the allocated frame produced by sip_rtp_read's
call of ast_dsp_process; free memory in parse unit tests
* func_dialgroup: properly deref ao2 object grhead in nominal path of
dialgroup_read
* func_odbc: free resultset in off nominal paths of odbc_read
* cli: free match_list in off nominal paths of CLI match completion
* config: free comment_buffer/list_buffer when configuration file load
is unchanged; free the same buffers any time they were
created and config files were processed
* data: free XML nodes in various places
* enum: free context buffer in off nominal paths
* features: free ast_call_feature in off nominal paths of applicationmap
config processing
* netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct
that is allocated by the method. Failures in
ast_sockaddr_resolve could result in the users of the method
not knowing whether or not the buffer was allocated. The
method will now not allocate the ast_sockaddr struct if it
will return failure.
* pbx: cleanup hash table traversals in off nominal paths; free
ignore pattern buffer if it already exists for the specified
context
* xmldoc: cleanup various nodes when we no longer need them
* main/editline: various cleanup of pointers not being freed before being
assigned to other memory, cleanup along off nominal paths
* menuselect/mxml: cleanup of value buffer for an attribute when that attribute
did not specify a value
* res_calendar*: responses are allocated via the various *_request method
returns and should not be allocated in the various
write_event methods; ensure attendee buffer is freed if no
data exists in the parsed node; ensure that calendar objects
are de-ref'd appropriately
* res_jabber: free buffer in off nominal path
* res_musiconhold: close the DIR* object in off nominal paths
* res_rtp_asterisk: if we run out of ports, close the rtp socket object and free
the rtp object
* res_srtp: if we fail to create the session in libsrtp, destroy the
temporary ast_srtp object
Jonathan Rose [Thu, 17 May 2012 14:41:13 +0000 (14:41 +0000)]
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.
........
(issue AST-876)
Merged revisions 366791 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Thu, 17 May 2012 12:57:30 +0000 (12:57 +0000)]
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
........
Merged revisions 366740 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Tue, 15 May 2012 20:44:59 +0000 (20:44 +0000)]
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.
Mark Michelson [Mon, 14 May 2012 20:06:58 +0000 (20:06 +0000)]
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
........
Merged revisions 366409 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Mon, 14 May 2012 19:16:36 +0000 (19:16 +0000)]
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
Russell Bryant [Fri, 11 May 2012 23:59:35 +0000 (23:59 +0000)]
format_mp3: Fix a possible crash in 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
........
Merged revisions 366296 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Thu, 10 May 2012 20:54:08 +0000 (20:54 +0000)]
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)
........
Merged revisions 366167 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Wed, 9 May 2012 19:12:32 +0000 (19:12 +0000)]
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.
Mark Michelson [Wed, 9 May 2012 16:15:28 +0000 (16:15 +0000)]
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
........
Merged revisions 365896 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Tue, 8 May 2012 20:25:08 +0000 (20:25 +0000)]
* 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.
........
Merged revisions 365692 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Tue, 8 May 2012 18:08:01 +0000 (18:08 +0000)]
* 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.
........
Merged revisions 365631 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Tue, 8 May 2012 15:51:13 +0000 (15:51 +0000)]
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
........
Merged revisions 365574 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Mon, 7 May 2012 18:39:10 +0000 (18:39 +0000)]
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.
Kinsey Moore [Fri, 4 May 2012 22:15:05 +0000 (22:15 +0000)]
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)
........
Merged revisions 365398 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Fri, 4 May 2012 16:28:06 +0000 (16:28 +0000)]
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/
........
Merged revisions 365313 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Fri, 4 May 2012 15:51:04 +0000 (15:51 +0000)]
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)
........
Merged revisions 365298 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Fix a CEL LINKEDID_END race and local channel linkedids
This patch has the ;2 channel inherit the linkedid of the ;1 channel and fixes
the race condition by no longer scanning the channel list for "other" channels
with the same linkedid. Instead, cel.c has an ao2 container of linkedid strings
and uses the refcount of the string as a counter of how many channels with the
linkedid exist. Not only does this eliminate the race condition, but it also
allows us to look up the linkedid by the hashed key instead of traversing the
entire channel list.
Don't leak a ref if out of memory and can't link the linkedid
If the ao2_link fails, we are most likely out of memory and bad things
are going to happen. Before those bad things happen, make sure to clean
up the linkedid references.
This patch also adds a comment explaining why linkedid can't be passed
to both local channel allocations and combines two ao2_ref calls into 1.
The security events framework API was changed in Asterisk 10 but the unit tests
were not updated at the same time.
This patch does the following:
* Adds two more security events that were added to the API
* Add challenge, received_challenge and received_hash in the inval_password
security event unit test
(issue ASTERISK-19760)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
issue-asterisk-19760-branch10.diff uploaded by Michael L. Young (license 5026)
Matthew Jordan [Wed, 2 May 2012 02:44:15 +0000 (02:44 +0000)]
Only log a failure to get read/write samples from factories if it didn't happen
In audiohook_read_frame_both, anytime samples are obtained from the read/write
factories a debug statement is logged stating that samples were not obtained
from the factories. This statement used to only occur if option_debug was
turned on and no samples were obtained; in some refactoring when the
option_debug statement was removed, the "else" clause was removed as well.
This patch makes it so that those debug log statements only occur if the
condition leading up to them actually happened.
Mark Michelson [Tue, 1 May 2012 23:10:16 +0000 (23:10 +0000)]
Fix Coverity-reported ARRAY_VS_SINGLETON error.
As it turned out, this wasn't a huge deal. We were calling
ast_app_parse_options() for a set of options of which none
took arguments. The proper thing to do for this case is to
pass NULL for the "args" parameter here. We were instead passing
a seemingly-randomly chosen char * from the function. While this
would never get written to, you can rest assured things would
have gotten bad had new options (which took arguments) been added
to func_volume.
(closes issue ASTERISK-19656)
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Merged revisions 364899 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Tue, 1 May 2012 19:07:09 +0000 (19:07 +0000)]
Play conf-placeintoconf message to the correct channel
Correct the code in app_confbridge to play the conf-placeintoconf message to
the marked user entering the bridge instead of to the conference while the
marked user hears silence.
(closes issue ASTERISK-19641) Reported-by: Mark A Walters
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Merged revisions 364786 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Mon, 30 Apr 2012 19:42:35 +0000 (19:42 +0000)]
Revert improved identities sent in dialog-info NOTIFY requests in r360862
Revision 360862 was intended to improve identities sent in dialog-info
NOTIFY requests. Some users reported that hint became broken once this
was done. It's not clear exactly what part of the patch has caused this
regression, but broken hints are bad.
For now, this revision is being reverted so that the next releases of
Asterisk do not have bad behavior in them. The original reported issue
will have to be fixed differently in the next version of Asterisk.
(issue ASTERISK-16735)
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Merged revisions 364706 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Sun, 29 Apr 2012 19:43:53 +0000 (19:43 +0000)]
Fix error that caused truncate operations to fail
Another very inappropriate placement of a ')' (again introduced in r362151)
caused the various truncate operations to attempt to truncate the sound file
at a position of '0'.
Michael L. Young [Sun, 29 Apr 2012 02:21:10 +0000 (02:21 +0000)]
Fix configuring custom sound_leader_has_left in confbridge.conf
The configuration option to specify a custom sound_leader_has_left file for a
conference bridge was not being parsed. This patch fixes it so that a custom
sound file will now be used.
(closes issue ASTERISK-19771)
Reported by: Pawel Kuzak
Tested by: Pawel Kuzak, Michael L. Young
Patches: leaderhasleft_sound.dpatch uploaded by Pawel Kuzak (license 6380)
Terry Wilson [Fri, 27 Apr 2012 22:31:01 +0000 (22:31 +0000)]
Fix ast_parse_arg numeric type range checking and add tests
ast_parse_arg wasn't checking for strto* parse errors or limiting
the results by the actual range of the numeric types. This patch fixes
that and adds unit tests as well.
Matthew Jordan [Fri, 27 Apr 2012 19:30:19 +0000 (19:30 +0000)]
Prevent overflow in calculation in ast_tvdiff_ms on 32-bit machines
The method ast_tvdiff_ms attempts to calculate the difference, in milliseconds,
between two timeval structs, and return the difference in a 64-bit integer.
Unfortunately, it assumes that the long tv_sec/tv_usec members in the timeval
struct are large enough to hold the calculated values before it returns. On
64-bit machines, this might be the case, as a long may be 64-bits. On 32-bit
machines, however, a long may be less (32-bits), in which case, the calculation
can overflow.
This overflow caused significant problems in MixMonitor, which uses the method
to determine if an audio factory, which has not presented audio to an audiohook,
is merely late in providing said audio or will never provide audio. In an
overflow situation, the audiohook would incorrectly determine that an audio
factory that will never provide audio is merely late instead. This led to
situations where a MixMonitor never recorded any audio. Note that this happened
most frequently when that MixMonitor was started by the ConfBridge application
itself, or when the MixMonitor was attached to a Local channel.
(issue ASTERISK-19497)
Reported by: Ben Klang
Tested by: Ben Klang
Patches:
32-bit-time-overflow-10-2012-04-26.diff (license #6283) by mjordan
(closes issue ASTERISK-19727)
Reported by: Mark Murawski
Tested by: Michael L. Young
Patches:
32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan)
Kinsey Moore [Fri, 27 Apr 2012 18:58:34 +0000 (18:58 +0000)]
Allow SIP pvts involved in Replaces transfers to fall out of reference sooner
Unref the SIP pvt stored in the refer structure as soon as it is no longer
needed so that the pvt and associated file descriptors can be freed sooner.
This change makes a reference decrement unnecessary in code that handles SIP
BYE/Also transfers which should not touch the reference anyway.
(Closes issue ASTERISK-19579)
Reported by: Maciej Krajewski
Tested by: Maciej Krajewski
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Merged revisions 364258 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Fri, 27 Apr 2012 14:44:13 +0000 (14:44 +0000)]
Allow for reloading SRTP crypto keys within the same SIP dialog
As a continuation of the patch in r356604, which allowed for the
reloading of SRTP keys in re-INVITE transfer scenarios, this patch
addresses the more common case where a new key is requested within
the context of a current SIP dialog. This can occur, for example, when
certain phones request a SIP hold.
Previously, once a dialog was associated with an SRTP object, any
subsequent attempt to process crypto keys in any SDP offer - either
the current one or a new offer in a new SIP request - were ignored. This
patch changes this behavior to only ignore subsequent crypto keys within
the current SDP offer, but allows future SDP offers to change the keys.
(issue ASTERISK-19253)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Richard Mudgett [Thu, 26 Apr 2012 20:25:05 +0000 (20:25 +0000)]
Fix DTMF atxfer running h exten after the wrong bridge ends.
When party B does an attended transfer of party A to party C, the
attending bridge between party B and C should not be running an h exten
when the bridge ends. Running an h exten now sets a softhangup flag to
ensure that an AGI will run in dead AGI mode.
* Set the AST_FLAG_BRIDGE_HANGUP_DONT on the party B channel for the
attending bridge between party B and C.
The reference held for SIP blind transfers using the Replaces header in an
INVITE was never freed on success and also failed to be freed in some error
conditions. This caused a file descriptor leak since the RTP structures in use
at the time of the transfer were never freed. This reference leak and another
relating to subscriptions in the same code path have now been corrected.
(Closes issue ASTERISK-19579)
Reported by: Maciej Krajewski
Tested by: Maciej Karjewski
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Merged revisions 363986 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Wed, 25 Apr 2012 20:48:22 +0000 (20:48 +0000)]
Make DAHDISendCallreroutingFacility wait 5 seconds for a reply before disconnecting the call.
Some switches may not handle the call-deflection/call-rerouting message if
the call is disconnected too soon after being sent. Asteisk was not
waiting for any reply before disconnecting the call.
* Added a 5 second delay before disconnecting the call to wait for a
potential response if the peer does not disconnect first.
(closes issue ASTERISK-19708)
Reported by: mehdi Shirazi
Patches:
jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett
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Merged revisions 363730 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Wed, 25 Apr 2012 19:47:44 +0000 (19:47 +0000)]
Clear ISDN channel resetting state if the peer continues to use it.
Some ISDN switches occasionally fail to send a RESTART ACKNOWLEDGE in
response to a RESTART request.
* Made the second SETUP received after sending a RESTART request clear the
channel resetting state as if the peer had sent the expected RESTART
ACKNOWLEDGE before continuing to process the SETUP. The peer may not be
sending the expected RESTART ACKNOWLEDGE.
Richard Mudgett [Wed, 25 Apr 2012 01:23:08 +0000 (01:23 +0000)]
Fix recalled party B feature flags for a failed DTMF atxfer.
1) B calls A with Dial option T
2) B DTMF atxfer to C
3) B hangs up
4) C does not answer
5) B is called back
6) B answers
7) B cannot initiate transfers anymore
* Add dial features datastore to recalled party B channel that is a copy
of the original party B channel's dial features datastore.
* Extracted add_features_datastore() from add_features_datastores().
* Renamed struct ast_dial_features features_caller and features_callee
members to my_features and peer_features respectively. These better names
eliminate the need for some explanatory comments.
* Simplified code accessing the struct ast_dial_features datastore.
Jonathan Rose [Mon, 23 Apr 2012 14:39:48 +0000 (14:39 +0000)]
AST-2012-004: Fix an error that allows AMI users to run shell commands sans authorization.
As detailed in the advisory, AMI users without write authorization for SYSTEM class AMI
actions were able to run system commands by going through other AMI commands which did
not require that authorization. Specifically, GetVar and Status allowed users to do this
by setting their variable/s options to the SHELL or EVAL functions.
Also, within 1.8, 10, and trunk there was a similar flaw with the Originate action that
allowed users with originate permission to run MixMonitor and supply a shell command
in the Data argument. That flaw is fixed in those versions of this patch.
(closes issue ASTERISK-17465)
Reported By: David Woolley
Patches:
162_ami_readfunc_security_r2.diff uploaded by jrose (license 6182)
18_ami_readfunc_security_r2.diff uploaded by jrose (license 6182)
10_ami_readfunc_security_r2.diff uploaded by jrose (license 6182)
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Merged revisions 363117 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
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Merged revisions 363141 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Mon, 23 Apr 2012 14:07:29 +0000 (14:07 +0000)]
AST-2012-006: Fix crash in UPDATE handling when no channel owner exists
If Asterisk receives a SIP UPDATE request after a call has been terminated and
the channel has been destroyed but before the SIP dialog has been destroyed, a
condition exists where a connected line update would be attempted on a
non-existing channel. This would cause Asterisk to crash. The patch resolves
this by first ensuring that the SIP dialog has an owning channel before
attempting a connected line update. If an UPDATE request is received and no
channel is associated with the dialog, a 481 response is sent.
(closes issue ASTERISK-19770)
Reported by: Thomas Arimont
Tested by: Matt Jordan
Patches:
ASTERISK-19278-2012-04-16.diff uploaded by Matt Jordan (license 6283)
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Merged revisions 363106 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Mon, 23 Apr 2012 13:48:48 +0000 (13:48 +0000)]
Reference skinny_subchannel object instead of skinny_device for r363103
The check-in to resolve ASTERISK-19592 (r363103) failed to switch to the
skinny_subchannel object instead of the skinny_device when attempting to
reference the buffer for the keypad digits. This patch fixes that.
(issue ASTERISK-19592)
Reported by: Russell Bryant
Matthew Jordan [Mon, 23 Apr 2012 13:40:23 +0000 (13:40 +0000)]
AST-2012-005: Fix remotely exploitable heap overflow in keypad button handling
When handling a keypad button message event, the received digit is placed into
a fixed length buffer that acts as a queue. When a new message event is
received, the length of that buffer is not checked before placing the new digit
on the end of the queue. The situation exists where sufficient keypad button
message events would occur that would cause the buffer to be overrun. This
patch explicitly checks that there is sufficient room in the buffer before
appending a new digit.
(closes issue ASTERISK-19592)
Reported by: Russell Bryant
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Merged revisions 363100 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
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Merged revisions 363102 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Michael L. Young [Fri, 20 Apr 2012 16:47:51 +0000 (16:47 +0000)]
Add missing payload type to events API
The Security Events Framework API was changed while adding the generation of
security events in chan_sip. A payload type and name was missed from being
added to struct ie_maps.
(closes issue ASTERISK-19759)
Reported by: Michael L. Young
Patches:
issue-asterisk-19759.diff uploaded by Michael L. Young (license 5026)
Terry Wilson [Fri, 20 Apr 2012 14:49:42 +0000 (14:49 +0000)]
Document Speech* apps hangup on failure and suggest TryExec
The Speech API apps return -1 on failure, which will hang up the channel. This
may not be desirable behavior for some, but it isn't something that can be
changed without breaking people's dialplans or writing an option to all of the
Speech apps that does what TryExec already does. This patch documents the
hangup behavior of the apps, and suggests TryExec as the solution.
(closes issue AST-813)
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Merged revisions 362815 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Michael L. Young [Thu, 19 Apr 2012 21:11:35 +0000 (21:11 +0000)]
Add leading and trailing backslashes
A couple of unit tests did not have have leading or trailing backslashes when
setting their test category resulting in a warning message being displayed.
Added the backslash where needed.
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Merged revisions 362680 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Sean Bright [Thu, 19 Apr 2012 16:04:21 +0000 (16:04 +0000)]
Prevent a crash in ExternalIVR when the 'S' command is sent first.
If the first command sent from an ExternalIVR client is an 'S' command, we were
blindly removing the first element from the play list and deferencing it, even
if it was NULL. This corrects that and also locks appropriately in one place.
(issue ASTERISK-17889)
Reported by: Chris Maciejewski
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Merged revisions 362586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Terry Wilson [Thu, 19 Apr 2012 14:31:59 +0000 (14:31 +0000)]
Handle multiple commands per connection via netconsole
Asterisk would accept multiple NULL-delimited CLI commands via the
netconsole socket, but would occasionally miss a command due to the
command not being completely read into the buffer. This patch ensures
that any partial commands get moved to the front of the read buffer,
appended to, and properly sent.
Matthew Jordan [Thu, 19 Apr 2012 02:27:08 +0000 (02:27 +0000)]
Fix a variety of potential buffer overflows
* chan_mobile: Fixed an overrun where the cind_state buffer (an integer array
of size 16) would be overrun due to improper bounds checking. At worst, the
buffer can be overrun by a total of 48 bytes (assuming 4-byte integers),
which would still leave it within the allocated memory of struct hfp. This
would corrupt other elements in that struct but not necessarily cause any
further issues.
* app_sms: The array imsg is of size 250, while the array (ud) that the data
is copied into is of size 160. If the size of the inbound message is
greater then 160, up to 90 bytes could be overrun in ud. This would corrupt
the user data header (array udh) adjacent to ud.
* chan_unistim: A number of invalid memmoves are corrected. These would move
data (which may or may not be valid) into the ends of these buffers.
* asterisk: ast_console_toggle_loglevel does not check that the console log
level being set is less then or equal to the allowed log levels of 32.
* format_pref: In ast_codec_pref_prepend, if any occurrence of the specified
codec is not found, the value used to index into the array pref->order
would be one greater then the maximum size of the array.
* jitterbuf: If the element being placed into the jitter buffer lands in the
last available slot in the jitter history buffer, the insertion sort attempts
to move the last entry in the buffer into one slot past the maximum length
of the buffer. Note that this occurred for both the min and max jitter
history buffers.
* tdd: If a read from fsk_serial returns a character that is greater then 32,
an attempt to read past one of the statically defined arrays containing the
values that character maps to would occur.
* localtime: struct ast_time and tm are not the same size - ast_time is larger,
although it contains the elements of tm within it in the same layout. Hence,
when using memcpy to copy the contents of tm into ast_time, the size of tm
should be used, as opposed to the size of ast_time.
* extconf: this treats ast_timing's minmask array as if it had a length of 48,
when it has defined the size of the array as 24. pbx.h defines minmask as
having a size of 48.
(issue ASTERISK-19668)
Reported by: Matt Jordan
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Merged revisions 362485 from http://svn.asterisk.org/svn/asterisk/branches/1.8
projects / thirdparty / asterisk.git / log
