asterisk/asterisk.git
8 years agoFix checking bounds of array index after using it; improper sizeof
Matthew Jordan [Thu, 17 May 2012 13:21:19 +0000 (13:21 +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
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Merged revisions 366740 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 366741 from http://svn.asterisk.org/svn/asterisk/branches/10

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8 years agoRemove missed idx parameter to some ao2 global holder macros.
Richard Mudgett [Wed, 16 May 2012 18:00:18 +0000 (18:00 +0000)]
Remove missed idx parameter to some ao2 global holder macros.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366700 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoChange ao2 global array to ao2 global object holder.
Richard Mudgett [Wed, 16 May 2012 16:34:42 +0000 (16:34 +0000)]
Change ao2 global array to ao2 global object holder.

Review: https://reviewboard.asterisk.org/r/1921/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366663 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoCorrect misuse of ast_strip_quoted() when getting a Diversion header's reason parameter.
Mark Michelson [Tue, 15 May 2012 23:41:59 +0000 (23:41 +0000)]
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 366598 from http://svn.asterisk.org/svn/asterisk/branches/10

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8 years agoThe predial routine must be run on the local;1 channel.
Richard Mudgett [Tue, 15 May 2012 19:36:26 +0000 (19:36 +0000)]
The predial routine must be run on the local;1 channel.

When ast_call() operates on a local channel, it copies a lot of things
from the local;1 channel to the local;2 channel.  This includes among
other things, channel variables and party id information.

Other reasons it was a bad idea to run predial on the local;2 channel:

1) The channel has not been completely setup.  The ast_call() completes
the setup.

2) The local;2 caller and connected line party information is opposite to
any other channels predial runs on.  (And it hasn't been setup yet.)

* Partially back out -r366183 by removing the chan_local implementation of
the struct ast_channel_tech.pre_call callback.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366546 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd predial support to FollowMe.
Richard Mudgett [Tue, 15 May 2012 16:53:09 +0000 (16:53 +0000)]
Add predial support to FollowMe.

Like the new predial feature for Dial.  This adds the same b/B options to
FollowMe.

Review: https://reviewboard.asterisk.org/r/1910/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366507 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMake chan_local use the API call instead of inlining its own version.
Richard Mudgett [Mon, 14 May 2012 21:34:14 +0000 (21:34 +0000)]
Make chan_local use the API call instead of inlining its own version.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366462 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix two more coverity constant expression result findings.
Mark Michelson [Mon, 14 May 2012 20:15:33 +0000 (20:15 +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
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Merged revisions 366409 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 366412 from http://svn.asterisk.org/svn/asterisk/branches/10

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8 years agoCommit framework for HANGUPCAUSE (replacement for SIP_CAUSE)
Kinsey Moore [Mon, 14 May 2012 19:44:27 +0000 (19:44 +0000)]
Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)

This is the starting point for the Asterisk 11: Who Hung Up work and provides
a framework which will allow channel drivers to report the types of hangup
cause information available in SIP_CAUSE without incurring the overhead of the
MASTER_CHANNEL dialplan function. The initial implementation only includes
cause generation for chan_sip and does not include cause code translation
utilities.

This change deprecates SIP_CAUSE and replaces its method of reporting cause
codes with the new framework. This change also deprecates the 'storesipcause'
option in sip.conf.

Review: https://reviewboard.asterisk.org/r/1822/
(Closes issue SWP-4221)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix broken reinvite glare scenario.
Mark Michelson [Mon, 14 May 2012 19:27:58 +0000 (19:27 +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

Review: https://reviewboard.asterisk.org/r/1911
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Merged revisions 366390 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366401 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMacro AST_PKG_CONFIG_CHECK to use chkconfig
Tzafrir Cohen [Mon, 14 May 2012 13:42:49 +0000 (13:42 +0000)]
Macro AST_PKG_CONFIG_CHECK to use chkconfig

AST_PKG_CONFIG_CHECK: Similar to AST_EXT_LIB_CHECK, but simply uses
pkg-config data.

This simple version only uses pkg-config(1)'s tests.

This commit also uses the macro to test for GTK2 and GMIME (instead of
the current direct usage of pkg-config).

Review: https://reviewboard.asterisk.org/r/1906/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366351 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoformat_mp3: Fix a possible crash in mp3_read().
Russell Bryant [Sat, 12 May 2012 00:03:42 +0000 (00:03 +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
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Merged revisions 366296 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 366297 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366298 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years ago* Made ast_change_name() hold the channels container lock while changing the channel...
Richard Mudgett [Thu, 10 May 2012 23:49:07 +0000 (23:49 +0000)]
* 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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Merged revisions 366241 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366242 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoTweak app_dial predial documentation.
Richard Mudgett [Thu, 10 May 2012 21:38:12 +0000 (21:38 +0000)]
Tweak app_dial predial documentation.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366193 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRun predial routine on local;2 channel where you would expect.
Richard Mudgett [Thu, 10 May 2012 21:29:41 +0000 (21:29 +0000)]
Run predial routine on local;2 channel where you would expect.

Before this patch, the predial routine executes on the ;1 channel of a
local channel pair.  Executing predial on the ;1 channel of a local
channel pair is of limited utility.  Any channel variables set by the
predial routine executing on the ;1 channel will not be available when the
local channel executes dialplan on the ;2 channel.

* Create ast_pre_call() and an associated pre_call() technology callback
to handle running the predial routine.  If a channel technology does not
provide the callback, the predial routine is simply run on the channel.

Review: https://reviewboard.asterisk.org/r/1903/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366183 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoResolve FORWARD_NULL static analysis warnings
Kinsey Moore [Thu, 10 May 2012 20:56:09 +0000 (20:56 +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)
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Merged revisions 366167 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 366168 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366169 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoCoverity Report: Fix issues for error type CHECKED_RETURN for core
Jonathan Rose [Thu, 10 May 2012 18:35:14 +0000 (18:35 +0000)]
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366126 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoClose the proper tcptls_session when session creation fails.
Mark Michelson [Thu, 10 May 2012 16:22:36 +0000 (16:22 +0000)]
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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Merged revisions 366053 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366062 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoCoverity Report: Fix issues for error type UNINIT in Core supported modules
Jonathan Rose [Thu, 10 May 2012 15:57:26 +0000 (15:57 +0000)]
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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Merged revisions 366049 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366051 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoBlock on frameout if the hardware has enough samples to complete a frame.
Jonathan Rose [Wed, 9 May 2012 19:28:47 +0000 (19:28 +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.

(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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366007 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agopass BUILD_CFLGAS and BUILD_LDFLAGS to menuselect
Tzafrir Cohen [Wed, 9 May 2012 19:26:08 +0000 (19:26 +0000)]
pass BUILD_CFLGAS and BUILD_LDFLAGS to menuselect

Allow menuselect to get its set of CFLAGS and LDFLAGS through the
environment of Make:

  make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"

Review: https://reviewboard.asterisk.org/r/1907/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366002 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoImprove FollowMe accept/decline DTMF string matching.
Richard Mudgett [Wed, 9 May 2012 17:58:11 +0000 (17:58 +0000)]
Improve FollowMe accept/decline DTMF string matching.

If you hit the wrong DTMF digit trying to accept/decline a FollowMe call,
you had to wait for the prompt to repeat to try again.

* Make FollowMe compare the last DTMF digits received to the
accept/decline matching strings.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365951 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoPrevent sip_pvt refleak when an ast_channel outlasts its corresponding sip_pvt.
Mark Michelson [Wed, 9 May 2012 16:36:10 +0000 (16:36 +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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365913 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoKeep answered FollowMe calls until call accepted or last step times out.
Richard Mudgett [Wed, 9 May 2012 02:35:29 +0000 (02:35 +0000)]
Keep answered FollowMe calls until call accepted or last step times out.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365856 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoPut winning FollowMe outgoing call on hold if the caller put it on hold.
Richard Mudgett [Wed, 9 May 2012 01:59:14 +0000 (01:59 +0000)]
Put winning FollowMe outgoing call on hold if the caller put it on hold.

The FollowMe caller call leg is usually answered and listening to MOH.
The caller could put the call on hold while FollowMe is looking for a
winner.  The winning outgoing call is now immediately placed on hold if
the caller has put the call on hold before the winning call was selected.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365829 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRestructure how the FollowMe outgoing channel list is handled.
Richard Mudgett [Wed, 9 May 2012 01:36:07 +0000 (01:36 +0000)]
Restructure how the FollowMe outgoing channel list is handled.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365828 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAddendum to -r365766. Since it is no longer allocated.
Richard Mudgett [Tue, 8 May 2012 22:46:14 +0000 (22:46 +0000)]
Addendum to -r365766.  Since it is no longer allocated.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365790 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMake FollowMe findmeexec() put the list head on the stack instead of mallocing it.
Richard Mudgett [Tue, 8 May 2012 22:25:42 +0000 (22:25 +0000)]
Make FollowMe findmeexec() put the list head on the stack instead of mallocing it.

Why this tiny struct was malloced instead of the 28k struct in the last
change is beyond me.  Just doing my part to help stamp out sillyness.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365766 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd interrupt ('I') command to ExternalIVR.
Sean Bright [Tue, 8 May 2012 21:46:21 +0000 (21:46 +0000)]
Add interrupt ('I') command to ExternalIVR.

Sending the 'I' command from an external process will cause the current playlist
to be cleared, including stopping any audio file that is currently playing.  This
is useful when you want to interrupt audio playback only when specific DTMF is
entered by the caller.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365751 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMake FollowMe app_exec() not declare a 28k struct on the stack.
Richard Mudgett [Tue, 8 May 2012 21:41:58 +0000 (21:41 +0000)]
Make FollowMe app_exec() not declare a 28k struct on the stack.

Helping to stamp out stack abuse.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365749 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoSimplify findmeexec() parameter passing.
Richard Mudgett [Tue, 8 May 2012 21:15:58 +0000 (21:15 +0000)]
Simplify findmeexec() parameter passing.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365711 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years ago* Fix FollowMe memory leak on error paths in app_exec().
Richard Mudgett [Tue, 8 May 2012 20:32:11 +0000 (20:32 +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 fm_args.namerecloc[].
This fixes unexpected namerecloc filename length restriction.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365707 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years ago* Fix accept/decline DTMF buffer overwrite in FollowMe.
Richard Mudgett [Tue, 8 May 2012 18:16:04 +0000 (18:16 +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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365633 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoSend more accurate identification information in dialog-info SIP NOTIFYs.
Mark Michelson [Tue, 8 May 2012 15:57:14 +0000 (15:57 +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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365576 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoChange comment to use local channel name designators in features.c
Richard Mudgett [Mon, 7 May 2012 20:08:37 +0000 (20:08 +0000)]
Change comment to use local channel name designators in features.c

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365532 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix channel opaquification slip-up in r365477
Matthew Jordan [Mon, 7 May 2012 18:58:40 +0000 (18:58 +0000)]
Fix channel opaquification slip-up in r365477

Those channels are opaque now...

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365480 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix type punned compiler warning in test_config.c
Richard Mudgett [Mon, 7 May 2012 18:51:44 +0000 (18:51 +0000)]
Fix type punned compiler warning in test_config.c
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365479 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoSupport VoiceMail d() option when extension does not exist in channel's context
Matthew Jordan [Mon, 7 May 2012 18:42:48 +0000 (18:42 +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.

(closes issue ASTERISK-18243)
Reported by: mjordan
Tested by: mjordan

Review: https://reviewboard.asterisk.org/r/1892
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365477 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix many issues from the NULL_RETURNS Coverity report
Kinsey Moore [Fri, 4 May 2012 22:17:38 +0000 (22:17 +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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365400 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix local channel chains optimizing themselves out of a call.
Richard Mudgett [Fri, 4 May 2012 17:38:39 +0000 (17:38 +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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365356 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix core FINDING 2, FINDING 3, and FINDING 4 from Coverity's CONSTANT_EXPRESSION_RESU...
Mark Michelson [Fri, 4 May 2012 15:52:30 +0000 (15:52 +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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365300 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoUpdate security events unit tests
Michael L. Young [Thu, 3 May 2012 19:36:33 +0000 (19:36 +0000)]
Update security events unit tests

The security events framework API was changed in Asterisk 10 but the unit tests
were not updated at the same time.

This patch does the following:
* Adds two more security events that were added to the API
* Add challenge, received_challenge and received_hash in the inval_password
  security event unit test

(Closes issue ASTERISK-19760)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
issue-asterisk-19760-trunk.diff uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/1897/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365248 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoUpdate documentation references in CHANGES to reflect the correct pages on the wiki.
Sean Bright [Thu, 3 May 2012 18:43:54 +0000 (18:43 +0000)]
Update documentation references in CHANGES to reflect the correct pages on the wiki.

The current CHANGES file refers to doc/ in many places and those files no longer exist.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365213 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix warning of Coverity Static analysis, change H225ProtocolIdentifier
Alexandr Anikin [Thu, 3 May 2012 15:05:14 +0000 (15:05 +0000)]
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365161 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd IPv6 support to ExternalIVR.
Sean Bright [Thu, 3 May 2012 14:47:58 +0000 (14:47 +0000)]
Add IPv6 support to ExternalIVR.

Review: https://reviewboard.asterisk.org/r/1896/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365158 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix coverity static analysis warning, allocate full ie structure
Alexandr Anikin [Thu, 3 May 2012 14:35:30 +0000 (14:35 +0000)]
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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8 years agoMultiple revisions 365006,365068
Terry Wilson [Wed, 2 May 2012 17:43:16 +0000 (17:43 +0000)]
Multiple revisions 365006,365068

........
  r365006 | twilson | 2012-05-02 10:49:03 -0500 (Wed, 02 May 2012) | 12 lines

  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.

  Review: https://reviewboard.asterisk.org/r/1895/
........
  r365068 | twilson | 2012-05-02 12:02:39 -0500 (Wed, 02 May 2012) | 11 lines

  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.

  Review: https://reviewboard.asterisk.org/r/1895/
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8 years agoBlocked revisions 365014
Michael L. Young [Wed, 2 May 2012 16:17:34 +0000 (16:17 +0000)]
Blocked revisions 365014

........
Update security events unit tests

The security events framework API was changed in Asterisk 10 but the unit tests
were not updated at the same time.

This patch does the following:
* Adds two more security events that were added to the API
* Add challenge, received_challenge and received_hash in the inval_password
  security event unit test

(issue ASTERISK-19760)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
issue-asterisk-19760-branch10.diff uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/1877/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365016 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoSave the address on which a MESSAGE was received, so it can be used in MESSAGE()
Jason Parker [Wed, 2 May 2012 15:59:43 +0000 (15:59 +0000)]
Save the address on which a MESSAGE was received, so it can be used in MESSAGE()

This is useful in cases where chan_sip may be listening on multiple addresses.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365011 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoOnly log a failure to get read/write samples from factories if it didn't happen
Matthew Jordan [Wed, 2 May 2012 02:51:02 +0000 (02:51 +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.
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8 years agoRemove a function that has been marked unused since Asterisk 1.6.0.
Mark Michelson [Tue, 1 May 2012 23:23:44 +0000 (23:23 +0000)]
Remove a function that has been marked unused since Asterisk 1.6.0.

The reason I'm removing this is that Coverity reported a STRAY_SEMICOLON
issue here. Since the function has been unused for so long, I just elected
to remove it altogether.

(closes issue ASTERISK-19660)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364915 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFixed __ao2_ref() validating user_data twice.
Richard Mudgett [Tue, 1 May 2012 23:21:07 +0000 (23:21 +0000)]
Fixed __ao2_ref() validating user_data twice.

(closes issue ASTERISK-19755)
Reported by: Gunther Kelleter
Patches:
      ao2_ref.patch (license #6372) patch uploaded by Gunther Kelleter
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364910 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix Coverity-reported ARRAY_VS_SINGLETON error.
Mark Michelson [Tue, 1 May 2012 23:11:22 +0000 (23:11 +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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364901 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years ago* Fix error path resouce leak in local_request().
Richard Mudgett [Tue, 1 May 2012 22:00:11 +0000 (22:00 +0000)]
* Fix error path resouce leak in local_request().

* Restructure local_request() to reduce indentation.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364846 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoPrevent a potential crash when using manager hooks.
Jason Parker [Tue, 1 May 2012 21:49:25 +0000 (21:49 +0000)]
Prevent a potential crash when using manager hooks.

Found by me while poking at DPMA-127.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364844 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoPlay conf-placeintoconf message to the correct channel
Kinsey Moore [Tue, 1 May 2012 19:10:48 +0000 (19:10 +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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364788 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix bad check in voicemail functions for ast_inboxcount2_func
Jonathan Rose [Tue, 1 May 2012 18:29:58 +0000 (18:29 +0000)]
Fix bad check in voicemail functions for ast_inboxcount2_func

Check looks for ast_inboxcount_func instead of ast_inboxcount2_func on
ast_inboxcount2_func calls.

(closes issue ASTERISK-19718)
Reported by: Corey Farrell
Patches:
ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364785 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoRevert revision 360862.
Mark Michelson [Mon, 30 Apr 2012 19:51:55 +0000 (19:51 +0000)]
Revert revision 360862.

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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8 years agoMerged revisions 364635 via svnmerge from
Mark Murawki [Mon, 30 Apr 2012 17:17:51 +0000 (17:17 +0000)]
Merged revisions 364635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r364635 | markm | 2012-04-30 11:51:12 -0400 (Mon, 30 Apr 2012) | 10 lines

  Sanatize result from bfd_find_nearest_line (BETTER_BACKTRACES)

  bfd_find_nearest_line can possibly set file to null resulting in a crash when strrchr(file) runs

  (closes issue ASTERISK-19815)
  Reported by Mark Murawski
  Tested by Mark Murawski
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8 years agoFix use freed pointer in return value from call thread
Alexandr Anikin [Mon, 30 Apr 2012 16:59:53 +0000 (16:59 +0000)]
Fix use freed pointer in return value from call thread

(issue ASTERISK-19663)
Reported by: Matt Jordan
Patches:
  ASTERISK-19663-ooh323.patch (License #5415)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364652 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix error that caused truncate operations to fail
Matthew Jordan [Sun, 29 Apr 2012 19:50:57 +0000 (19:50 +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'.

(issue ASTERISK-19655)
Reported by: Matt Jordan

(issue ASTERISK-19810)
Reported by: colbec
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8 years agoFix configuring custom sound_leader_has_left in confbridge.conf
Michael L. Young [Sun, 29 Apr 2012 02:23:22 +0000 (02:23 +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)

Review: https://reviewboard.asterisk.org/r/1884/
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8 years agoAdd support for lightweight NAT keepalive.
Joshua Colp [Sat, 28 Apr 2012 20:24:45 +0000 (20:24 +0000)]
Add support for lightweight NAT keepalive.

If enabled using the keepalive option in sip.conf a small packet will be sent
at a regular interval to keep the NAT mapping open. This is lightweight as the
remote side does not need to parse and handle a SIP message.

(closes issue AST-783)
Review: https://reviewboard.asterisk.org/r/1756/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364500 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agomd5: supress some compiler warnings.
Russell Bryant [Sat, 28 Apr 2012 01:33:49 +0000 (01:33 +0000)]
md5: supress some compiler warnings.

md5.c: In function ‘MD5Final’:
md5.c:154:2: error: dereferencing type-punned pointer will break strict-aliasing rules [-Werror=strict-aliasing]
md5.c:155:2: error: dereferencing type-punned pointer will break strict-aliasing rules [-Werror=strict-aliasing]

There is an md5 unit test and it still passes.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364462 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agores_corosync: Fix build against corosync 2.0.
Russell Bryant [Sat, 28 Apr 2012 01:20:57 +0000 (01:20 +0000)]
res_corosync: Fix build against corosync 2.0.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364444 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoapp_minivm: Fix a couple compiler warnings.
Russell Bryant [Sat, 28 Apr 2012 01:10:35 +0000 (01:10 +0000)]
app_minivm: Fix a couple compiler warnings.

The warnings were about argv[0] being used uninitialized, which is correct.
Just remove setting username to this value, since username is set again before
it actually gets used.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364438 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agofeatures: Add FEATURE() and FEATUREMAP() functions.
Russell Bryant [Sat, 28 Apr 2012 00:58:54 +0000 (00:58 +0000)]
features: Add FEATURE() and FEATUREMAP() functions.

Add two new dialplan functions: FEATURE() and FEATUREMAP().  FEATURE()
lets you set some of the configuration options from the [general] section
of features.conf on a per-channel basis.  FEATUREMAP() lets you customize
the key sequence used to activate built-in features, such as blindxfer,
and automon.  See the built-in documentation for details.

Review: https://reviewboard.asterisk.org/r/1871/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364437 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoPreDial - Ability to run dialplan on callee and caller channels before Dial.
Richard Mudgett [Sat, 28 Apr 2012 00:31:47 +0000 (00:31 +0000)]
PreDial - Ability to run dialplan on callee and caller channels before Dial.

Thanks to Mark Murawski for the initial patch and feature definition.

(closes issue ASTERISK-19548)
Reported by: Mark Murawski

Review: https://reviewboard.asterisk.org/r/1878/
Review: https://reviewboard.asterisk.org/r/1229/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364436 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMultiple revisions 364365,364369
Terry Wilson [Fri, 27 Apr 2012 22:54:20 +0000 (22:54 +0000)]
Multiple revisions 364365,364369

........
  r364365 | twilson | 2012-04-27 17:31:01 -0500 (Fri, 27 Apr 2012) | 11 lines

  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.

  Review: https://reviewboard.asterisk.org/r/1879/
  ........

  Merged revisions 364340 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r364369 | twilson | 2012-04-27 17:33:10 -0500 (Fri, 27 Apr 2012) | 2 lines

  Add missing test_config.c
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8 years agoDon't attempt to make use of the dynamic_exclude_static ACL if DNS lookup fails.
Mark Michelson [Fri, 27 Apr 2012 22:11:01 +0000 (22:11 +0000)]
Don't attempt to make use of the dynamic_exclude_static ACL if DNS lookup fails.

(closes issue ASTERISK-18321)
Reported by Dan Lukes
Patches:
ASTERISK-18321.patch by Mark Michelson (license #5049)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364343 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoPrevent overflow in calculation in ast_tvdiff_ms on 32-bit machines
Matthew Jordan [Fri, 27 Apr 2012 19:30:59 +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)

(closes issue ASTERISK-19471)
Reported by: feyfre
Tested by: feyfre

(issue ASTERISK-19426)
Reported by: Johan Wilfer

Review: https://reviewboard.asterisk.org/r/1889/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364287 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAllow SIP pvts involved in Replaces transfers to fall out of reference sooner
Kinsey Moore [Fri, 27 Apr 2012 18:59:36 +0000 (18:59 +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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364260 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAllow for reloading SRTP crypto keys within the same SIP dialog
Matthew Jordan [Fri, 27 Apr 2012 14:45:08 +0000 (14:45 +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

Review: https://reviewboard.asteriskorg/r/1885/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364205 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agofix a wrong behavior of alarm timezones in caldav and icalendar when an alarm doesnt...
Stefan Schmidt [Fri, 27 Apr 2012 12:58:03 +0000 (12:58 +0000)]
fix a wrong behavior of alarm timezones in caldav and icalendar when an alarm doesnt use utc. This change uses the same timezone from the start time.
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8 years agoUpdate Pickup application documentation. (With feeling this time.)
Richard Mudgett [Thu, 26 Apr 2012 21:11:25 +0000 (21:11 +0000)]
Update Pickup application documentation. (With feeling this time.)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364110 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix DTMF atxfer running h exten after the wrong bridge ends.
Richard Mudgett [Thu, 26 Apr 2012 20:35:41 +0000 (20:35 +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.

(closes issue AST-870)

(closes issue ASTERISK-19717)
Reported by: Mario

(closes issue ASTERISK-19633)
Reported by: Andrey Solovyev
Patches:
      jira_asterisk_19633_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Andrey Solovyev, Mario
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364082 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoAdd more constness to the end_buf pointer in the netconsole
Terry Wilson [Thu, 26 Apr 2012 19:33:49 +0000 (19:33 +0000)]
Add more constness to the end_buf pointer in the netconsole

issue ASTERISK-18308
Review: https://reviewboard.asterisk.org/r/1876/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364048 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoCode formatting fixes.
Olle Johansson [Thu, 26 Apr 2012 13:59:11 +0000 (13:59 +0000)]
Code formatting fixes.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363989 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix reference leaks involving SIP Replaces transfers
Kinsey Moore [Thu, 26 Apr 2012 13:31:16 +0000 (13:31 +0000)]
Fix reference leaks involving SIP Replaces transfers

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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363988 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agochan_sip: [general] maxforwards, not checked for a value greater than 255
Alec L Davis [Thu, 26 Apr 2012 09:48:55 +0000 (09:48 +0000)]
chan_sip: [general] maxforwards, not checked for a value greater than 255

The peer maxforwards is checked for both '< 1' and '> 255',
but the default 'maxforwards' in the [general] section is only checked for '< 1'

alecdavis (license 585)
Reported by: alecdavis
Tested by: alecdavis

Review: https://reviewboard.asterisk.org/r/1888/
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Merged revisions 363935 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363936 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoUpdate Pickup application documentation. (Even better)
Richard Mudgett [Thu, 26 Apr 2012 03:12:44 +0000 (03:12 +0000)]
Update Pickup application documentation. (Even better)
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Merged revisions 363875 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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8 years ago* Put more information in pickup_exec() LOG_NOTICE.
Richard Mudgett [Thu, 26 Apr 2012 01:29:09 +0000 (01:29 +0000)]
* Put more information in pickup_exec() LOG_NOTICE.

* Delay duplicating a string on the stack in pickup_exec().

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363839 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoUpdate Pickup application documentation.
Richard Mudgett [Wed, 25 Apr 2012 23:00:26 +0000 (23:00 +0000)]
Update Pickup application documentation.
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8 years agoMake DAHDISendCallreroutingFacility wait 5 seconds for a reply before disconnecting...
Richard Mudgett [Wed, 25 Apr 2012 20:51:58 +0000 (20:51 +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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8 years agoClear ISDN channel resetting state if the peer continues to use it.
Richard Mudgett [Wed, 25 Apr 2012 19:55:12 +0000 (19:55 +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.

(issue ASTERISK-19608)
(issue AST-844)
(issue AST-815)
Patches:
      jira_ast_815_v1.8.patch (license #5621) patch uploaded by rmudgett (modified)
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8 years agoAdd documentation
Olle Johansson [Wed, 25 Apr 2012 13:57:01 +0000 (13:57 +0000)]
Add documentation

Thanks Tilghman!

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8 years agoFormatting changes only
Olle Johansson [Wed, 25 Apr 2012 11:18:14 +0000 (11:18 +0000)]
Formatting changes only

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363599 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoUse the DEFINED value for musicclass length.
Olle Johansson [Wed, 25 Apr 2012 10:49:13 +0000 (10:49 +0000)]
Use the DEFINED value for musicclass length.

For some reason, features.c has it's own definition. Should propably be fixed too.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363595 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoMake it possible to change the minimum DTMF duration in asterisk.conf
Olle Johansson [Wed, 25 Apr 2012 09:32:21 +0000 (09:32 +0000)]
Make it possible to change the minimum DTMF duration in asterisk.conf

Asterisk has a setting for the minimum allowed DTMF. If we get shorter
DTMF tones, these will be changed to the minimum on the outbound call
leg.

(closes issue ASTERISK-19772)

Review: https://reviewboard.asterisk.org/r/1882/
Reported by: oej
Tested by: oej
Patches by: oej

Thanks to the reviewers.

1.8 branch for this patch: agave-dtmf-duration-asterisk-conf-1.8

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363558 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFormatting fixes
Olle Johansson [Wed, 25 Apr 2012 08:39:01 +0000 (08:39 +0000)]
Formatting fixes

Developer guidelines are important.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363517 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFormatting fixes
Olle Johansson [Wed, 25 Apr 2012 08:02:52 +0000 (08:02 +0000)]
Formatting fixes

Found a small amount of curly brackets in my hotel room here in Denmark.
I hereby donate them to the Asterisk project.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363480 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoFix recalled party B feature flags for a failed DTMF atxfer.
Richard Mudgett [Wed, 25 Apr 2012 01:26:44 +0000 (01:26 +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.

(closes issue ASTERISK-19383)
Reported by: lgfsantos
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8 years agoHangup affected channel in error paths of bridge_call_thread().
Richard Mudgett [Wed, 25 Apr 2012 00:03:52 +0000 (00:03 +0000)]
Hangup affected channel in error paths of bridge_call_thread().
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8 years agoOpenBSD doesn't have rawmemchr, use strchr
Terry Wilson [Tue, 24 Apr 2012 17:52:26 +0000 (17:52 +0000)]
OpenBSD doesn't have rawmemchr, use strchr

(closes issue ASTERISK-19758)
Reported by: Barry Miller
Tested by: Terry Wilson
Patches:
  362758-diff uploaded by Barry Miller (license 5434)
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8 years agoMake app_dial and app_queue use new macro and gosub calls.
Richard Mudgett [Mon, 23 Apr 2012 17:05:55 +0000 (17:05 +0000)]
Make app_dial and app_queue use new macro and gosub calls.

* Simplify some code in app_dial and app_queue by calling
ast_app_exec_macro() and ast_app_exec_sub().

* Fix minor locking issue in app_dial for post-answer macro/gosub
MACRO/GOSUB_RESULT=GOTO: handling.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363269 65c4cc65-6c06-0410-ace0-fbb531ad65f3

8 years agoOn some platforms, O_RDONLY is not a flag to be checked, but merely the absence of...
Tilghman Lesher [Mon, 23 Apr 2012 16:08:33 +0000 (16:08 +0000)]
On some platforms, O_RDONLY is not a flag to be checked, but merely the absence of O_RDWR and O_WRONLY.

The POSIX specification does not mandate how these 3 flags must be specified,
only that one of the three must be specified in every call.
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8 years agoAST-2012-004: Fix an error that allows AMI users to run shell commands sans authoriza...
Jonathan Rose [Mon, 23 Apr 2012 14:48:22 +0000 (14:48 +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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8 years agoAST-2012-006: Fix crash in UPDATE handling when no channel owner exists
Matthew Jordan [Mon, 23 Apr 2012 14:10:19 +0000 (14:10 +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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8 years agoAST-2012-005: Fix remotely exploitable heap overflow in keypad button handling
Matthew Jordan [Mon, 23 Apr 2012 13:53:24 +0000 (13:53 +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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8 years agores_corosync: Recover if corosync gets restarted.
Russell Bryant [Sat, 21 Apr 2012 11:45:28 +0000 (11:45 +0000)]
res_corosync: Recover if corosync gets restarted.

If corosync gets restarted while Asterisk is running, automatically recover.

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