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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363877 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* 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

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363790 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363740 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363689 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Thanks Tilghman!


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363637 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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

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

Developer guidelines are important.


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

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

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363377 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363335 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* 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

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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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363159 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363108 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363105 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363046 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Reimplement the "corosync show members" CLI command using a CPG iterator
instead of the cpg_membership_get API call.  This will also show all
CPG members, including those in groups other than 'asterisk', which may
be useful at some point for debugging purposes.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363045 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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* Redo ast_app_run_sub()/ast_app_exec_sub() to use a known return point so
execution will stop after the routine returns there.
(s@gosub_virtual_context:1)

* Create ast_app_exec_macro() and ast_app_exec_sub() to run the macro and
gosub application respectively with the parameter string already created.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362962 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Move debug message in ast_rtp_instance_early_bridge_make_compatible() to
be output when what it states has actually happened.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362920 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362919 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The CHANNEL_DEADLOCK_AVOIDANCE() feature of preserving where the channel
lock was originally obtained is overkill where ast_channel_lock_both() was
inlined.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362888 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Simplify some dialplan priority setting code in ast_explicit_goto()
because of opaquification.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362867 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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ISDN ETSI PTP and Q.SIG (And SS7 in future) have support for reporting who
was the original redirecting party of a call.

* Added support for the original redirecting party and reason to the
REDIRECTING function and the system core as well as to the stubbed
locations in sig_pri.c.

Review: https://reviewboard.asterisk.org/r/1829/


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362731 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362679 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1732/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362635 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362588 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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.

(closes issue ASTERISK-18308)
Review: https://reviewboard.asterisk.org/r/1876/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362538 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362497 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The Security Events Framework API changed in trunk to support IPv6.  This broke
the building of the security events test which was based around IPv4.  This
patches fixes the build by changing the test to conform to the new changes.

(related to issue ASTERISK-19447)

Review: https://reviewboard.asterisk.org/r/1874/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362432 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Several telcos bring the BRI PTMP layer 1 down when the line is idle.
When layer 1 goes down, Asterisk cannot make outgoing calls.  Incoming
calls could fail as well because the alarm processing is handled by a
different code path than the Q.931 messages.

* Add the layer1_presence configuration option to ignore layer 1 alarms
when the telco brings layer 1 down.  This option can be configured by span
while the similar DAHDI driver teignorered=1 option is system wide.  This
option unlike layer2_persistence does not require libpri v1.4.13 or newer.

Related to JIRA AST-598

JIRA ABE-2845
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362430 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In ast_codec_pref_getsize, if an unknown format is passed to the method,
no preferred codec will be selected and a negative number will be used to
index into the format list.  The method now logs an unknown format as a
warning, and returns an empty format list.

(issue ASTERISK-19655)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1863/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362380 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch addresses a number of modules in resources that did not handle the
negative return value from function calls adequately.  This includes:

* res_agi.c: if the result of the read function is a negative number,
indicating some failure, the result would instead be treated as the number
of bytes read.  This patch now treats negative results in the same manner
as an end of file condition, with the exception that it also logs the
error code indicated by the return.

* res_musiconhold.c: if spawn_mp3 fails to assign a file descriptor to srcfd,
and instead assigns a negative value, that file descriptor could later be
passed to functions that require a valid file descriptor.  If spawn_mp3 fails,
we now immediately retry instead of continuing in the logic.

* res_rtp_asterisk.c: if no codec can be matched between two RTP instances
in a peer to peer bridge, we immediately return instead of attempting to
use the codec payload type as an index to determine the appropriate negotiated
codec.

(issue ASTERISK-19655)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1863/
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