Multiple revisions 361208,361211,361270,361330,361333,361381,361422,361472,361522...

Multiple revisions 361208,361211,361270,361330,361333,361381,361422,361472,361522,361560,361607,361658,361706,361754,361804,361855,361907,361956,361981,362080,362084,362152,362202,362205,362264,362305,362356-362357,362360,362364,362377,362429,362496,362537,362587,362678,362681,362730,362816,362869,362918,362998,363212,363376,363429,363688,363734,363789,363876,363935,363987,364047,364065,364109,364163,364204,364259,364285,364342,364365,364369,364536,364579,364650-364651,364777,364787,364845,364900,364903,364965,365014,365083

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  r361208 | jrose | 2012-04-04 14:30:09 -0500 (Wed, 04 Apr 2012) | 10 lines

  Make 'help devstate change' display properly (get rid of excess comma)

  (closes issue ASTERISK-19444)
  Reported by: Makoto Dei
  Patches:
  	devstate-change-usage-truncate.patch uploaded by Makoto Dei (license 5027)
  ........

  Merged revisions 361201 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r361211 | jrose | 2012-04-04 15:00:23 -0500 (Wed, 04 Apr 2012) | 12 lines

  Fix some stuff involving calls to memcpy and memset

  The important parts of the patch were already applied through other updates.

  (closes issue ASTERISK-19445)
  Reported by: Makoto Dei
  Patches:
  	memset-memcpy-length.patch uploaded by Makoto Dei (license 5027)
  ........

  Merged revisions 361210 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r361270 | jrose | 2012-04-05 11:53:35 -0500 (Thu, 05 Apr 2012) | 10 lines

  Fix MusicOnHold in MeetMe so that it always uses the class if it's been defined

  There were a few instances of restarting music on hold in meetme that would cause
  Asterisk to revert to the default class of music on hold for no adequate reason.

  Review: https://reviewboard.asterisk.org/r/1844/
  ........

  Merged revisions 361269 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r361330 | kmoore | 2012-04-06 08:31:51 -0500 (Fri, 06 Apr 2012) | 11 lines

  Remove unnecessary error message in app_dial.c

  The error message for failure to stop autoservice after a gosub or macro call
  during a dial was removed for macro while Asterisk 1.4 was still being actively
  developed. The corresponding gosub error message was never removed.

  (closes issue ASTERISK-19551)
  ........

  Merged revisions 361329 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r361333 | mjordan | 2012-04-06 09:01:33 -0500 (Fri, 06 Apr 2012) | 11 lines

  Fix a typo in the warning messages for an ignored media stream

  Added a '\n' to the warning messages when we ignore a media stream due to the
  port number being '0'.

  (closes issue ASTERISK-19646)
  Reported by: Badalian Vyacheslav
  ........

  Merged revisions 361332 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r361381 | russell | 2012-04-06 10:49:19 -0500 (Fri, 06 Apr 2012) | 5 lines

  Remove a few more files related to chan_usbradio and app_rpt.
  ........

  Merged revisions 361380 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r361422 | pabelanger | 2012-04-06 11:31:18 -0500 (Fri, 06 Apr 2012) | 14 lines

  Multiple revisions 361403,361412

  ........
    r361403 | pabelanger | 2012-04-06 12:24:36 -0400 (Fri, 06 Apr 2012) | 2 lines

    Fix typo in svn:keywords
  ........
    r361412 | pabelanger | 2012-04-06 12:27:30 -0400 (Fri, 06 Apr 2012) | 2 lines

    Fix typo in svn:keywords
  ........

  Merged revisions 361403,361412 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r361472 | kmoore | 2012-04-06 13:13:04 -0500 (Fri, 06 Apr 2012) | 5 lines

  Add missing newlines to CLI logging
  ........

  Merged revisions 361471 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r361522 | rmudgett | 2012-04-06 14:47:29 -0500 (Fri, 06 Apr 2012) | 8 lines

  Don't add an empty MESSAGE_DATA(key) header if it doesn't already exist.

  Doing Set(MESSAGE_DATA(key)=) would add an empty key header if the key
  header did not already exist.  If it already existed it would delete it.

  * Made msg_set_var_full() exit early if the named variable did not already
  exist and the value to set is empty.
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  r361560 | mjordan | 2012-04-06 15:32:13 -0500 (Fri, 06 Apr 2012) | 13 lines

  Fix memory leak when using MeetMeAdmin 'e' option with user specified

  A memory leak/reference counting leak occurs if the MeetMeAdmin 'e' command
  (eject last user that joined) is used in conjunction with a specified user.
  Regardless of the command being executed, if a user is specified for the
  command, MeetMeAdmin will look up that user.  Because the 'e' option kicks
  the last user that joined, as opposed to the one specified, the reference to
  the user specified by the command would be leaked when the user variable
  was assigned to the last user that joined.
  ........

  Merged revisions 361558 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r361607 | mjordan | 2012-04-06 17:00:11 -0500 (Fri, 06 Apr 2012) | 12 lines

  Fix memory leak in res_calendar_ews when event email address node is empty

  If the XML calendar data returned by a Microsoft Exchange Web Service
  specifies an XML Event E-Mail Address ("EmailAddress"), and no e-mail address
  is provided, a condition existed where an ast_calendar_attendee struct would
  be allocated but not appended to the list of attendees.  Because of that,
  the memory associated with the attendee would never be freed.  This patch
  frees the memory if no e-mail address is provided.
  ........

  Merged revisions 361606 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r361658 | mjordan | 2012-04-09 14:42:53 -0500 (Mon, 09 Apr 2012) | 15 lines

  Change SHARED function to use a safe traversal when modifying a variable

  When the SHARED function modifies a variable, it removes it from its list of
  variables and reinserts the new value at the head of the list of variables.
  Doing this inside a standard list traversal can be dangerous, as the
  standard list traversal does not account for the list being changed.  While
  the code in question should not cause a use after free violation due to its
  breaking out of the loop after freeing the variable, it could lead to a
  maintenance issue if the loop was modified.  This also fixes a violation
  reported by a static analysis tool, which also makes this code easier to
  maintain in the future.
  ........

  Merged revisions 361657 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r361706 | mjordan | 2012-04-09 15:54:55 -0500 (Mon, 09 Apr 2012) | 17 lines

  Prevent invalid access of free'd memory if DAHDI channel during an MWI event

  In the MWI processing loop, when a valid event occurs the temporary caller ID
  information is deallocated.  If a new DAHDI channel is successfully created,
  the event is passed up to the analog_ss_thread without error and the loop
  exits.  If, however, the DAHDI channel is not created, then the caller ID
  struct has been free'd, and the gains reset to their previous level.  This
  will almost certainly cause an invalid access to the free'd memory, either
  in subsequent calls to callerid_free or calls to callerid_feed.

  This patch makes it so that we only free the caller ID structure if a
  DAHDI channel is successfully created, and we bump the gains back up
  if we fail to make a DAHDI channel.
  ........

  Merged revisions 361705 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r361754 | mjordan | 2012-04-09 16:44:30 -0500 (Mon, 09 Apr 2012) | 12 lines

  Allow func_curl to exit gracefully if list allocation fails during write

  If the global_curl_info data structure could not be allocated, the
  datastore associated with the operation would be free'd, but the function
  would not return.  This would later dereference the datastore, almost
  certainly causing Asterisk to crash.  With this patch, if the data
  structure is not allocated the method will return an error code, and
  not attempt any further operation.
  ........

  Merged revisions 361753 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r361804 | mjordan | 2012-04-10 14:57:30 -0500 (Tue, 10 Apr 2012) | 10 lines

  Fix crash caused by unloading or reloading of res_http_post

  When unlinking itself from the registered HTTP URIs, res_http_post could
  inadvertently free all URIs registered with the HTTP server.  This patch
  modifies the unregister method to only free the URI that is actually
  being unregistered, as opposed to all of them.
  ........

  Merged revisions 361803 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r361855 | rmudgett | 2012-04-10 16:47:42 -0500 (Tue, 10 Apr 2012) | 19 lines

  Prevent invalid access of free'd memory if DAHDI channel during an MWI event

  In the MWI processing loop, when a valid event occurs the temporary caller ID
  information is deallocated.  If a new DAHDI channel is successfully created,
  the event is passed up to the analog_ss_thread without error and the loop
  exits.  If, however, the DAHDI channel is not created, then the caller ID
  struct has been free'd, and the gains reset to their previous level.  This
  will almost certainly cause an invalid access to the free'd memory, either
  in subsequent calls to callerid_free or calls to callerid_feed.

  * Rework the -r361705 patch to better manage the cs and mtd allocated
  resources.

  * Fixed use of mwimonitoractive flag to be correct if the mwi_thread()
  fails to start.
  ........

  Merged revisions 361854 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r361907 | jrose | 2012-04-11 11:07:50 -0500 (Wed, 11 Apr 2012) | 10 lines

  Change default value of 'ignorebusy' on Queue members so that behavior is more like 1.8

  Prior to this patch, in order to restore that behavior, a function would have
  to be used on the QueueMember to make the ringinuse option do anything, which
  is pretty unreasonable.

  (closes issue ASTERISK-19536)
  reported by: Philippe Lindheimer
  Review: https://reviewboard.asterisk.org/r/1860/
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  r361956 | kmoore | 2012-04-12 10:01:13 -0500 (Thu, 12 Apr 2012) | 13 lines

  Simplify build system architecture optimization

  This change to the build system rips out any usage of PROC along with
  architecture-specific optimizations in favor of using -march=native where it is
  supported.  This fixes broken builds on 64bit Intel systems and results in
  better optimized code on systems running GCC 4.2+.

  Review: https://reviewboard.asterisk.org/r/1852/
  (closes issue ASTERISK-19462)
  ........

  Merged revisions 361955 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r361981 | kmoore | 2012-04-12 11:22:28 -0500 (Thu, 12 Apr 2012) | 12 lines

  Make trunkfreq take effect when set

  Previously, setting trunkfreq had no effect on initial load or on reload and
  only ever used the default value.  This causes trunkfreq to be used
  appropriately on initial load and reload.

  (closes issue ASTERISK-19521)
  Patch-by: Jaco Kroon
  ........

  Merged revisions 361972 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r362080 | jrose | 2012-04-13 10:30:22 -0500 (Fri, 13 Apr 2012) | 10 lines

  Send relative path named recordings to the meetme directory instead of sounds

  Prior to this patch, no effort was made to parse the path name to determine a proper
  destination for recordings of MeetMe's r option. This fixes that.

  Review: https://reviewboard.asterisk.org/r/1846/
  ........

  Merged revisions 362079 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r362084 | jrose | 2012-04-13 11:04:22 -0500 (Fri, 13 Apr 2012) | 15 lines

  Make ForkCDR e option not set end time of the newly forked CDR log

  Prior to this patch, ForkCDR's e option would immediately set the end time of the forked
  CDR to that of the CDR that is being terminated. This resulted in the new CDR's end time
  being roughly the same as it's beginning time (which is in turn roughly the same as the
  original's end time).

  (closes issue ASTERISK-19164)
  Reported by: Steve Davies
  Patches:
  	cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012)
  ........

  Merged revisions 362082 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r362152 | mjordan | 2012-04-16 14:39:32 -0500 (Mon, 16 Apr 2012) | 19 lines

  Check for IO stream failures in various format's truncate/seek operations

  For the formats that support seek and/or truncate operations, many of
  the C library calls used to determine or set the current position indicator
  in the file stream were not being checked.  In some situations, if an error
  occurred, a negative value would be returned from the library call.  This
  could then be interpreted inappropriately as positional data.

  This patch checks the return values from these library calls before
  using them in subsequent operations.

  (issue ASTERISK-19655)
  Reported by: Matt Jordan

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

  Merged revisions 362151 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r362202 | mjordan | 2012-04-16 16:40:29 -0500 (Mon, 16 Apr 2012) | 18 lines

  Fix handling of negative return code when storing voicemails in ODBC storage

  When storing a voicemail message using an ODBC connection to a database, the
  voicemail message is first stored on disk.  The sound file associated with
  the message is read into memory before being transmitted to the database.
  When this occurs, a failure in the C library's lseek function would cause a
  negative value to be passed to the mmap as the size of the memory map to
  create.  This would almost certainly cause the creation of the memory map to
  fail, resulting in the message being lost.

  (issue ASTERISK-19655)
  Reported by: Matt Jordan

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

  Merged revisions 362201 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r362205 | mjordan | 2012-04-16 16:57:19 -0500 (Mon, 16 Apr 2012) | 25 lines

  Fix negative return handling in channel drivers

  In chan_agent, while handling a channel indicate, the agent channel driver
  must obtain a lock on both the agent channel, as well as the channel the
  agent channel is using.  To do so, it attempts to lock the other channel
  first, then unlock the agent channel which is locked prior to entry into
  the indicate handler.  If this unlock fails with a negative return value,
  which can occur if the object passed to agent_indicate is an invalid ao2
  object or is NULL, the return value is passed directly to strerror, which
  can only accept positive integer values.

  In chan_dahdi, the return value of dahdi_get_index is used to directly
  index into the sub-channel array.  If dahd_get_index returns a negative
  value, it would use that value to index into the array, which could cause
  an invalid memory access.  If dahdi_get_index returns a negative number,
  we now default to SUB_REAL.

  (issue ASTERISK-19655)
  Reported by: Matt Jordan

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

  Merged revisions 362204 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r362264 | elguero | 2012-04-17 09:53:04 -0500 (Tue, 17 Apr 2012) | 23 lines

  Turn off warning message when bind address is set to any.

  When a bind address is set to an ANY address (udpbindport=::), a warning message
  is displayed stating that "Address remapping activated in sip.conf but we're
  using IPv6, which doesn't need it.  Please remove 'localnet' and/or 'externaddr'
  settings."  But if one is running dual stack, we shouldn't be told to turn those
  settings off.

  This patch checks if the bind address is an ANY address or not.  The warning
  message will now only be displayed if the bind address is NOT an ANY address and
  IPv6 is being used.

  Also, updated the copyright year.

  (closes issue ASTERISK-19456)
  Reported by: Michael L. Young
  Tested by: Michael L. Young
  Patches:
    chan_sip_ipv6_message.diff uploaded by Michael L. Young (license 5026)
  ........

  Merged revisions 362253 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r362305 | mjordan | 2012-04-17 13:27:44 -0500 (Tue, 17 Apr 2012) | 15 lines

  Fix error that caused seek format operations to set max file size to '1' or '0'

  A very inappropriate placement of a ')' (introduced in r362151) caused the
  maximum size of a file to be set as the result of a comparison operation, as
  opposed to the result of the ftello operation.  This resulted in seeking being
  restricted to the beginning of the file, or 1 byte into the file.  Thanks to
  the Asterisk Test Suite for properly freaking out about this on at least one
  test.

  (issue ASTERISK-19655)
  Reported by: Matt Jordan
  ........

  Merged revisions 362304 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r362356 | mjordan | 2012-04-17 15:56:05 -0500 (Tue, 17 Apr 2012) | 17 lines

  Fix places where a negative return from ftello could be used as invalid input

  In a variety of locations in both reading and writing a file, the result
  from the C library function ftello is used as input to other functions.  For
  the parameters and functions in question, a negative value is invalid input.
  This patch checks the return value from the ftello function to determine if
  we were able to determine the current position in the file stream and, if not,
  fail gracefully.

  (issue ASTERISK-19655)
  Reported by: Matt Jordan

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

  Merged revisions 362355 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r362357 | jrose | 2012-04-17 15:57:36 -0500 (Tue, 17 Apr 2012) | 12 lines

  Make use of va_args more appropriate to form in various res_config modules plus utils.

  A number of va_copy operations weren't matched with a corresponding va_end in res_config_odbc. Also, there was a potential for va_end to be invoked twice on the same va_arg in utils, which would mean invoking va_end on an undefined variable... which is bad.
  va_end is removed from various functions in config_pgsql and config_curl since they aren't making their own copy.  The invokers of those functions are responsible for calling va_end on them.

  (issue ASTERISK-19451)
  Reported by: Walter Doekes
  Review: https://reviewboard.asterisk.org/r/1848/
  ........

  Merged revisions 362354 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r362360 | mjordan | 2012-04-17 16:07:29 -0500 (Tue, 17 Apr 2012) | 24 lines

  Fix places in main where a negative return value could impact execution

  This patch addresses a number of modules in main that did not handle the
  negative return value from function calls adequately, or were not sufficiently
  clear that the conditions leading to improper handling of the return values
  could not occur.  This includes:

  * asterisk.c: A negative return value from the read function would be used
  directly as an index into a buffer.  We now check for success of the read
  function prior to using its result as an index.

  * manager.c: Check for failures in mkstemp and lseek when handling the
  temporary file created for processing data returned from a CLI command in
  action_command.  Also check that the result of an lseek is sanitized prior
  to using it as the size of a memory map to allocate.

  (issue ASTERISK-19655)
  Reported by: Matt Jordan

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

  Merged revisions 362359 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r362364 | mjordan | 2012-04-17 16:11:25 -0500 (Tue, 17 Apr 2012) | 29 lines

  Fix places in resources where a negative return value could impact execution

  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/
  ........

  Merged revisions 362362 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r362377 | mjordan | 2012-04-17 16:22:37 -0500 (Tue, 17 Apr 2012) | 13 lines

  Handle case where an unknown format is used to get the preferred codec size

  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/
........
  r362429 | rmudgett | 2012-04-18 11:27:51 -0500 (Wed, 18 Apr 2012) | 19 lines

  Add ability to ignore layer 1 alarms for BRI PTMP lines.

  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
  ........

  Merged revisions 362428 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r362496 | mjordan | 2012-04-18 21:27:08 -0500 (Wed, 18 Apr 2012) | 50 lines

  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
  ........

  Merged revisions 362485 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r362537 | twilson | 2012-04-19 09:31:59 -0500 (Thu, 19 Apr 2012) | 14 lines

  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.

  (closes issue ASTERISK-18308)
  Review: https://reviewboard.asterisk.org/r/1876/
  ........

  Merged revisions 362536 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r362587 | seanbright | 2012-04-19 11:04:21 -0500 (Thu, 19 Apr 2012) | 12 lines

  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
  ........

  Merged revisions 362586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r362678 | rmudgett | 2012-04-19 16:00:21 -0500 (Thu, 19 Apr 2012) | 5 lines

  Update membermacro and membergosub documentation in queues.conf.sample.
  ........

  Merged revisions 362677 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r362681 | elguero | 2012-04-19 16:11:35 -0500 (Thu, 19 Apr 2012) | 9 lines

  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.
  ........

  Merged revisions 362680 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r362730 | wdoekes | 2012-04-19 16:59:43 -0500 (Thu, 19 Apr 2012) | 5 lines

  Fix documentation for ${VERSION(ASTERISK_VERSION_NUM)}.
  ........

  Merged revisions 362729 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r362816 | twilson | 2012-04-20 09:49:42 -0500 (Fri, 20 Apr 2012) | 13 lines

  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)
  ........

  Merged revisions 362815 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r362869 | twilson | 2012-04-20 11:12:34 -0500 (Fri, 20 Apr 2012) | 11 lines

  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)
  ........

  Merged revisions 362868 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r362918 | elguero | 2012-04-20 11:47:51 -0500 (Fri, 20 Apr 2012) | 11 lines

  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)
........
  r362998 | rmudgett | 2012-04-20 20:45:13 -0500 (Fri, 20 Apr 2012) | 5 lines

  Update app_dial M and U option GOTO return value documentation.
  ........

  Merged revisions 362997 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r363212 | tilghman | 2012-04-23 11:06:53 -0500 (Mon, 23 Apr 2012) | 8 lines

  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.
  ........

  Merged revisions 363209 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r363376 | rmudgett | 2012-04-24 19:01:21 -0500 (Tue, 24 Apr 2012) | 5 lines

  Hangup affected channel in error paths of bridge_call_thread().
  ........

  Merged revisions 363375 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r363429 | rmudgett | 2012-04-24 20:23:08 -0500 (Tue, 24 Apr 2012) | 27 lines

  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
  ........

  Merged revisions 363428 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r363688 | rmudgett | 2012-04-25 14:47:44 -0500 (Wed, 25 Apr 2012) | 19 lines

  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)
  ........

  Merged revisions 363687 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r363734 | rmudgett | 2012-04-25 15:48:22 -0500 (Wed, 25 Apr 2012) | 18 lines

  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
  ........

  Merged revisions 363730 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r363789 | rmudgett | 2012-04-25 17:59:46 -0500 (Wed, 25 Apr 2012) | 5 lines

  Update Pickup application documentation.
  ........

  Merged revisions 363788 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r363876 | rmudgett | 2012-04-25 22:11:45 -0500 (Wed, 25 Apr 2012) | 5 lines

  Update Pickup application documentation. (Even better)
  ........

  Merged revisions 363875 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r363935 | alecdavis | 2012-04-26 04:46:38 -0500 (Thu, 26 Apr 2012) | 14 lines

  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/
  ........

  Merged revisions 363934 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r363987 | kmoore | 2012-04-26 08:27:34 -0500 (Thu, 26 Apr 2012) | 15 lines

  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)
  Reported by: Maciej Krajewski
  Tested by: Maciej Karjewski
  ........

  Merged revisions 363986 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r364047 | twilson | 2012-04-26 14:30:55 -0500 (Thu, 26 Apr 2012) | 8 lines

  Add more constness to the end_buf pointer in the netconsole

  issue ASTERISK-18308
  Review: https://reviewboard.asterisk.org/r/1876/
  ........

  Merged revisions 364046 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r364065 | rmudgett | 2012-04-26 15:25:05 -0500 (Thu, 26 Apr 2012) | 24 lines

  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
  ........

  Merged revisions 364060 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r364109 | rmudgett | 2012-04-26 16:10:46 -0500 (Thu, 26 Apr 2012) | 5 lines

  Update Pickup application documentation. (With feeling this time.)
  ........

  Merged revisions 364108 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r364163 | schmidts | 2012-04-27 07:54:19 -0500 (Fri, 27 Apr 2012) | 3 lines

  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.
........
  r364204 | mjordan | 2012-04-27 09:44:13 -0500 (Fri, 27 Apr 2012) | 23 lines

  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/
  ........

  Merged revisions 364203 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r364259 | kmoore | 2012-04-27 13:58:34 -0500 (Fri, 27 Apr 2012) | 14 lines

  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
  ........

  Merged revisions 364258 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r364285 | mjordan | 2012-04-27 14:30:19 -0500 (Fri, 27 Apr 2012) | 43 lines

  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/
  ........

  Merged revisions 364277 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r364342 | mmichelson | 2012-04-27 16:58:06 -0500 (Fri, 27 Apr 2012) | 10 lines

  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)
  ........

  Merged revisions 364341 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  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
........
  r364369 | twilson | 2012-04-27 17:33:10 -0500 (Fri, 27 Apr 2012) | 2 lines

  Add missing test_config.c
........
  r364536 | elguero | 2012-04-28 21:21:10 -0500 (Sat, 28 Apr 2012) | 13 lines

  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/
........
  r364579 | mjordan | 2012-04-29 14:43:53 -0500 (Sun, 29 Apr 2012) | 15 lines

  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
  ........

  Merged revisions 364578 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r364650 | markm | 2012-04-30 11:43:11 -0500 (Mon, 30 Apr 2012) | 15 lines

  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
  ........
........
  r364651 | may | 2012-04-30 11:48:57 -0500 (Mon, 30 Apr 2012) | 10 lines

  Fix use freed pointer in return value from call thread

  (issue ASTERISK-19663)
  Reported by: Matt Jordan
  Patches:
    ASTERISK-19663-ooh323.patch (License #5415)
  ........

  Merged revisions 364649 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r364777 | jrose | 2012-05-01 13:23:08 -0500 (Tue, 01 May 2012) | 13 lines

  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)
  ........

  Merged revisions 364769 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r364787 | kmoore | 2012-05-01 14:07:09 -0500 (Tue, 01 May 2012) | 12 lines

  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
  ........

  Merged revisions 364786 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r364845 | rmudgett | 2012-05-01 16:50:32 -0500 (Tue, 01 May 2012) | 7 lines

  * Fix error path resouce leak in local_request().

  * Restructure local_request() to reduce indentation.
  ........

  Merged revisions 364840 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r364900 | mmichelson | 2012-05-01 18:10:16 -0500 (Tue, 01 May 2012) | 16 lines

  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)
  ........

  Merged revisions 364899 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r364903 | rmudgett | 2012-05-01 18:14:12 -0500 (Tue, 01 May 2012) | 10 lines

  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
  ........

  Merged revisions 364902 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r364965 | mjordan | 2012-05-01 21:44:15 -0500 (Tue, 01 May 2012) | 11 lines

  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.
........
  r365014 | elguero | 2012-05-02 11:16:03 -0500 (Wed, 02 May 2012) | 18 lines

  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/
........
  r365083 | twilson | 2012-05-02 12:29:54 -0500 (Wed, 02 May 2012) | 33 lines

  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/
  ........

  Merged revisions 365006,365068 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 361208,361211,361270,361330,361333,361381,361422,361472,361522,361560,361607,361658,361706,361754,361804,361855,361907,361956,361981,362080,362084,362152,362202,362205,362264,362305,362356-362357,362360,362364,362377,362429,362496,362537,362587,362678,362681,362730,362816,362869,362918,362998,363212,363376,363429,363688,363734,363789,363876,363935,363987,364047,364065,364109,364163,364204,364259,364285,364342,364365,364369,364536,364579,364650-364651,364777,364787,364845,364900,364903,364965,365014,365083 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@365264 65c4cc65-6c06-0410-ace0-fbb531ad65f3