https://origsvn.digium.com/svn/asterisk/branches/10

........
  r346349 | dvossel | 2011-11-28 18:00:11 -0600 (Mon, 28 Nov 2011) | 10 lines
  
  Fixes memory leak in message API.
  
  The ast_msg_get_var function did not properly decrement
  the ref count of the var it retrieves.  The way this is
  implemented is a bit tricky, as we must decrement the var and then
  return the var's value.  As long as the documentation for the
  function is followed, this will not result in a dangling pointer as
  the ast_msg structure owns its own reference to the var while it
  exists in the var container.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346350 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-18693)
Reported by: Davide Dal Fra

Review: https://reviewboard.asterisk.org/r/1600/
Reviewed by: Walter Doekes
........

Merged revisions 346292 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 346293 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346294 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 346239 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 346240 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346241 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In r116240, get_msg_text() got an extra parameter to fix the unwanted
addition of trailing newlines to SIP MESSAGE bodies. This caused all
linefeeds to be trimmed, which isn't right either. This is a stop-gap;
the right fix is to return the original SIP request body.

Review: https://reviewboard.asterisk.org/r/1586
Reviewed by: Matt Jordan
........

Merged revisions 346147 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 346198 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346199 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1594
........

Merged revisions 346144 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 346145 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346146 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This should fix almost all resource leaks in res_jabber that involve
ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where
ast_aji_get_client would sometimes bump an object's refcount and sometimes not.

Review: https://reviewboard.asterisk.org/r/1553
........

Merged revisions 346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 346087 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346088 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When using the MessageSend application to send a SIP MESSAGE to a non-peer,
chan_sip attempted to validate the hostname or IP Address.  In the process,
it stripped off the extension and failed to add it back to the sip_pvt
structure before transmitting.  This patch adds the full URI passed in
from the message core to the sip_pvt structure.

(closes issue ASTERISK-18903)
Reported by: Shaun Clark
Tested by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1597/
........

Merged revisions 346040 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346053 65c4cc65-6c06-0410-ace0-fbb531ad65f3

As a result of the fix for ASTERISK-18039, realtime caching MOH no longer
properly resumes playing back a file between different holds in the same call.
This is because scanning for new files causes the existing file array to be
emptied and we were just comparing that the saved pointer to the filename
matched the pointer to the filename in a particular position in the array. An
easy fix is to save the filename instead of a pointer to it and then do a
strcmp instead of comparing the addresses.

(closes issue ASTERISK-18912)
Review: https://reviewboard.asterisk.org/r/1596/
........

Merged revisions 346030 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 346031 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346033 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 346029 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346032 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The dnsmgr refresh would always get the first address found regardless of
the original address family requested.  So if you asked for only IPv4
addresses originally, you might get an IPv6 address on refresh.

* Saved the original address family requested by ast_dnsmgr_lookup() to be
used when the address is refreshed.
........

Merged revisions 345976 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 345977 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345978 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-17973)
........

Merged revisions 345923 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 345924 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345925 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-18895)
Reported by: zvision
Patches:
        conf_config_parser.diff (license #5755) patch uploaded by zvision
........

Merged revisions 345882 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345883 65c4cc65-6c06-0410-ace0-fbb531ad65f3

It is possible to enumerate SIP usernames when the general and user/peer
nat settings differ in whether to respond to the port a request is sent
from or the port listed for responses in the Via header. In 1.4 and 1.6.2,
this would mean if one setting was nat=yes or nat=route and the other was
either nat=no or nat=never. In 1.8 and 10, this would mean when one was
nat=force_rport and the other was nat=no.

In order to address this problem, it was decided to switch the default
behavior to nat=yes/force_rport as it is the most commonly used option
and to strongly discourage setting nat per-peer/user when at all possible.

For more discussion of the issue, please see:
  http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html

(closes issue ASTERISK-18862)
Review: https://reviewboard.asterisk.org/r/1591/
........

Merged revisions 345776 from http://svn.asterisk.org/svn/asterisk/branches/1.4
........

Merged revisions 345800 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
........

Merged revisions 345828 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 345830 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345831 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This provides the same functionality as #include however an asterisk module will
still load if the filename does not exist.

Review: https://reviewboard.asterisk.org/r/1476/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345735 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1593/
........

Merged revisions 345682 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 345683 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345684 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In 1.8 and previous versions, one could use any fullword portion of the key
name, including the full key, to obtain the record.  Until this patch, this
did not work for the full key.

Closes issue ASTERISK-18886

Patch: by tilghman
Review: by twilson (http://pastebin.com/7rtu6bpk) on #asterisk-dev
........

Merged revisions 345640 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345643 65c4cc65-6c06-0410-ace0-fbb531ad65f3

in r342871

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345601 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds two new menu features to app_confbridge, admin_toggle_menu_
participants and participant_count.  The admin action will globally mute /
unmute all non-admin participants on a converence, while the participant
count simply exposes the existing participant count function to the
conference bridge menu.

This also adds configuration options to change the sound played when the
conference is globally muted / unmuted, as well as the necessary config
hooks to place these functions in the DTMF menus.

(closes issue ASTERISK-18204)
Reported by: Kevin Reeves
Tested by: Matt Jordan
Patches:
  app_confbridge.c.patch.txt, conf_config_parser.c.patch.txt, 
  confbridge.h.patch.txt uploaded by Kevin Reeves (license 6281)

Review: https://reviewboard.asterisk.org/r/1518/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345560 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Dead code makes programmers sick.  I am sick of looking at it.
........

Merged revisions 345546 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 345558 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345559 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Before, using the U flag in Voicemail with multiple recipients would put urgent messages
in the INBOX folder for all users past the first thanks to a bug with the message
copying function. This would also cause messages to fail to be sent if the INBOX
directory hadn't been created for that mailbox yet.

(closes issue ASTERISK-18245)
Reported by: Matt Jordan

(closes issue ASTERISK-18246)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1589/
........

Merged revisions 345487 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 345488 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345489 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Change from using send() to ast_agi_send() so the HANGUP shows up in the
AGI debug output.

(closes issue ASTERISK-18723)
Reported by: James Van Vleet
Patches:
      jira_asterisk_18723_v1.8.patch (license #5621) patch uploaded by rmudgett
........

Merged revisions 345431 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 345432 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345433 65c4cc65-6c06-0410-ace0-fbb531ad65f3

It is fortunate that the typo does not alter generated code since the
e->restart.channel and e->ring.channel members are in the same position.

(closes issue ASTERISK-18868)
Reported by: zvision
Patches:
      sig_pri.c.diff (License #5755) patch uploaded by zvision
........

Merged revisions 345370 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 345371 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345375 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Add parameter to queue log ADDMEMBER to indicate if the member is
paused.

(closes issue ASTERISK-18645)
Reported by: garlew
Patches:
      paused.diff (License #5337) patch uploaded by garlew
Tested by: rmudgett, garlew

Review: https://reviewboard.asterisk.org/r/1469/
........

Merged revisions 345285 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 345290 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345317 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support
working correctly removed a long standing ability to do overlap dialing
using DTMF in the early media phase of a call.

See ASTERISK-18702 it has a very good description of the issue.

I started with Pavel Troller's chan_sip.diff patch on issue
ASTERISK-18702.

* Added 'dtmf' enum value to sip.conf allowoverlap config option.  The new
option value causes the Incomplte application to not send anything with
chan_sip so the caller can supply more digits via DTMF.

* Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE
since that is what it really means.

* Fixed get_destination() inconsistency with the pickup extension
matching.

* Fixed initialization of PAGE3 of global_flags in reload_config().

(closes issue ASTERISK-18702)
Reported by: Pavel Troller

Review: https://reviewboard.asterisk.org/r/1517/

Review: https://reviewboard.asterisk.org/r/1582/
........

Merged revisions 345273 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 345275 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345276 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-18857)
Reported by: David M
Patches:
      mainpbx-trivial.patch (License #6326) patch uploaded by David M
........

Merged revisions 345219 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 345220 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345221 65c4cc65-6c06-0410-ace0-fbb531ad65f3

int blah = 1;
...
write(chan->alertpipe[1], &blah, new_frames * sizeof(blah)) !=
(new_frames * sizeof(blah)))

is only valid when new_frames == 1. Otherwise we start reading into adjacent
variables declared on the stack. The read end discards what is read, so the
values don't matter but it's not a good idea to read past where we want even
though new_frames is almost always 1 and should never be large. This patch is
basically taken out of kpfleming's eventfd branch, as he mentioned that he
remembered fixing it there when I talked to him about this issue.

Review: https://reviewboard.asterisk.org/r/1583/
........

Merged revisions 345163 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 345164 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345165 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The issue mentioned in the bug report had been fixed recently by
twilson. The reporter included this documentation fix.

(closes issue ASTERISK-18572)
Reported by: Richard Miller
Patch by: Richard Miller (modified)
........

Merged revisions 345160 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 345161 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345162 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change was made because forcegreeting and forcename settings in voicemail could be
circumvented by hanging up after entering a password, because the only way voicemail
currently observes whether a mailbox is new or not is by checking to see if the password
is the same as the mailbox number or not.

(closes issue ASTERISK-18282)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1581/
........

Merged revisions 345062 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 345117 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345120 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When sip_send_mwi_to_peer was modified recently to avoid deadlocks, vmexten
was not expected to be null.  This change handles that situation to avoid
a segfault.
........

Merged revisions 345063 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 345064 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345065 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345023 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prevent channels been parsed repetitively.
........

Merged revisions 344965 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 344966 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344979 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Realtime MOH class caching was implemented because without it, you would build
a completely new MOH class and would start the music over at the beginning each
time hold was pressed in a conversation. Unfortunately, this broke re-scanning
for file changes for realtime MOH classes. This patch corrects that issue.

(closes issue ASTERISK-18039)
Review: https://reviewboard.asterisk.org/r/1579/
........

Merged revisions 344899 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 344900 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344901 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Kevin P Fleming suggested that r343157 should use __alignof__ instead
of sizeof. For most systems this won't be an issue, but better fix it
now while it's still fresh.

Review: https://reviewboard.asterisk.org/r/1573
........

Merged revisions 344843 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 344845 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344846 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes the format type check in ast_closestream and 
filestream_destructor.  Previously a comparison operator was used, but since
audio formats are no longer contiguous (and AST_FORMAT_AUDIO_MASK includes
formats that have a value greater than the video formats), a bitwise AND
operation is used instead.  Duplicated code was also moved to filestream_close.

(closes issue ASTERISK-18682)
Reported by: Aldo Bedrij
Tested by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1580/
........

Merged revisions 344823 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 344842 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344844 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Nick Lewis added them in https://reviewboard.asterisk.org/r/549/diff/1-2/
for no apparent reason. There is no way that params could become NULL in
that piece of code, so I removed these excess checks again.
........

Merged revisions 344837 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 344839 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344840 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The opaque_data was added and enclosed in single quotes, assuming it
would be only a single line. The rest of the lines were appended after
the closing quote.

(closes issue ASTERISK-18852)
Reported by: peep_ on IRC

Review: https://reviewboard.asterisk.org/r/1577
........

Merged revisions 344835 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 344836 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344838 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If capability is adjusted when switching to UDPTL during fax transmission, fax
teardown fails.  Make sure capability is only touched if RTP is active.  This
regression was introduced in R344385.
........

Merged revisions 344769 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 344770 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344771 65c4cc65-6c06-0410-ace0-fbb531ad65f3

JIRA AST-710
........

Merged revisions 344715 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 344716 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344717 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Holding the channel lock while the CLI "core show channel" command is
executing can slow down the system.  It could block the system if the
console output is halted or paused.

* Made capture the CLI "core show channel" output into a buffer to be
output after the channel is unlocked.

* Removed use of C++ keyword as a variable name.  out renamed to obuf.

* Checked allocation of obuf for failure so will not crash.

(closes issue ASTERISK-18571)
Reported by: Pavel Troller
Tested by: rmudgett
........

Merged revisions 344661 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 344662 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344663 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Attempting to call an extension which used Caller ID matching with a channel that
has an empty caller id string would result in a segmentation fault.

(closes issue ASTERISK-18392
Reported By: Ales Zelenik
........

Merged revisions 344608 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 344609 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344610 65c4cc65-6c06-0410-ace0-fbb531ad65f3