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/
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(closes issue ASTERISK-18857)
Reported by: David M
Patches:
      mainpbx-trivial.patch (License #6326) patch uploaded by David M
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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/
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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)
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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/
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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.
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Prevent channels been parsed repetitively.
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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/
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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
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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/
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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.
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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
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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.
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JIRA AST-710
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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
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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
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(closes issue ASTERISK-18848)
Reported by: Tony Mountifield
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Fixed app_queue.c:ring_entry() calling ast_call() with the channel locks
held.  Chan_local attempts to do deadlock avoidance in its ast_call()
callback and could deadlock if a channel lock is already held.
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It was strange that the AgentCalled AMI event would get most of its
information from the incoming channel but then get the CallerID
information from the outgoing channel.  Before connected line support was
added, this information was always the same at this point.

(closes issue ASTERISK-18152)
Reported by: Thomas Farnham
Tested by: rmudgett
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https://origsvn.digium.com/svn/asterisk/branches/10

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  r344493 | dvossel | 2011-11-10 15:54:42 -0600 (Thu, 10 Nov 2011) | 12 lines
  
  Fixes issue with ConfBridge participants hanging up during DTMF feature menu usage getting stuck in conference forever.
  
  When a conference user enters the DTMF menu they are suspended from the
  bridge while the channel is handed off to the DTMF feature code.  If a
  user entered this state and hungup, there existed a race condition where
  the channel could not exit the conference because it was waiting on a
  signal that would never arrive.  This patch fixes that, because it would
  stupid for me to talk about the problem and commit a patch for something else.
  
  (closes issue ASTERISK-18829)
  Reported by: zvision
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344494 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This fixes an issue where a user of a dynamic conference was asked for a PIN
twice.  This also adds documentation to assist in future modifications to the
piece of code responsible for PIN checking.

(closes issue AST-670)
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Fix bug ASTERISK-16558 which dealt with the order of responses to incoming
streams defined by SDP.

Fix unreported bug where offering multiple same-type streams would cause
Asterisk to reply with an incorrect SDP response missing one or more streams
without a proper declination.

Fix bugs related to a single non-audio stream being offered with responses
requesting codecs that were not offered in the initial invite along with an
additional audio stream that was not in the initial invite.

Review: https://reviewboard.asterisk.org/r/1516/
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Patch by: jkonieczny (modified)
ASTERISK-18490
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Another deadlock between the conlock/hints and channels/channel locking
orders.

* Don't hold the channel and private lock in sip_new() when calling
ast_exists_extension().

(closes issue ASTERISK-18740)
Reported by: Byron Clark
Patches:
      sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by Gregory Hinton Nietsky
      ASTERISK-18740.patch (license #6157) patch uploaded by Byron Clark
Tested by: Byron Clark
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The domain matching code prior to 1.8 used to manually remove the port
from the host:port string when determining if an incoming request
matched the list of domains. When switching to the new parsing
functions, the documentation implied that the "domain" was being
returned by these functions, when instead it was returning the
"hostport" as defined by RFC 3261. This led to confusion and resulted
in 1.8+ rejecting an incoming request from x.x.x.x:xxxxx when
domain=x.x.x.x was set in sip.conf.

This patch renames the "domain" variables in the parsing functions to
"hostport" to more accurately describe what it is that they are
returning and also properly truncates the resulting hostport strings
when dealing with domain matching.

Review: https://reviewboard.asterisk.org/r/1574/
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Review: https://reviewboard.asterisk.org/r/1575/
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Some PBXes require this for call status checking

(closes issue ASTERISK-18748)
Reported by: Fabrizio Lazzaretti
Patches:
      ASTERISK-18748-5.patch (License #5415) patch uploaded by may213
Tested by: Fabrizio Lazzaretti
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The last time this code was touched (by me), a subtlety was missed based on the
difference between needing to check a pin's validity and the need to prompt
for a pin.

(closes issue ASTERISK-18488)
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ASTERISK-18739
Patch by: pawel (modified)
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Residual changes for Asterisk v10 branch from ASTERISK-18747 after
https://reviewboard.asterisk.org/r/1564/ commit and associated dialogs
callid hash key change fix.

* Make check_rtp_timeout() return CMP_MATCH if need to delete dialog from
dialogs_rtpcheck.  This is an optimization to avoid an unneeded
lock/unlock and object search when using ao2_unlink.

* Prevent crash in check_rtp_timeout() if dialog->rtp is NULL.

Review: https://reviewboard.asterisk.org/r/1557/
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"dialplan remove include x from y" crashed when the amount of arguments
was less than 6.

(closes issue ASTERISK-18762)
Reported by: Andrey Solovyev
Tested by: Andrey Solovyev
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https://origsvn.digium.com/svn/asterisk/branches/10

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  r343900 | dvossel | 2011-11-08 12:29:33 -0600 (Tue, 08 Nov 2011) | 11 lines
  
  Fixes regression caused by r343635
  
  There was a missing unlock for a function return that is only
  present in Asterisk 10 and Asterisk Trunk.
  
  (closes issue ASTERISK-18839)
  Reported by: Michael L. Young
  Patches:
      asterisk-18839-missing-lock-trunk-v2.diff (License #5026) patch uploaded by Michael L. Young
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343905 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fixed a LOG_ERROR message referencing the config variable list v that
had previously been processed and became NULL.

* Added error return value set that was missing in an ast_append_ha()
error return path.

(closes issue ASTERISK-18743)
Reported by: Michele
Patches:
      issueA18743-fix_dynamic_exclude_static_bad_host_log.patch (license #5674) patch uploaded by Walter Doekes
Tested by: Michele
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A hardcoded a branch number was in the prep_tarball which could not work. Changed
it to the  variable.
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The "Trust RPID" and "Send RPID" entries in the "sip show settings" CLI command
pulled the flags from the incorrect global flags page.  These are now read from
sip global flags page 0.

(closes issue AST-711)
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You can now use the built in variables , , and 
within a dynamic weight. For example, this could be useful when you want
to pass requested lookup number to the SHELL() function which could be
used to execute a script to dynamically set the weight of the result.

(Closes issue ASTERISK-13657)
Reported by: Joel Vandal
Tested by: Leif Madsen, Russell Bryant
Patches:
     asterisk-1.6-dundi-varhead.patch uploaded by Joel Vandal (License #5374)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343693 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTERISK-18669
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Changing an object value used as a container key requires removing the
object from the container and reinserting it.

* Created change_callid_pvt() to call instead of build_callid_pvt().  The
change_callid_pvt() will correctly change the dialog callid so the ao2
conainter can explicitly unlink it.
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Fix a locking inversion in sip_send_mwi_to_peer that was causing deadlocks.
This function now requires that both the peer and associated pvt be unlocked
before it is called for cases where peer and peer->mwipvt form a circular
reference.

(closes issue ASTERISK-18663)
Review: https://reviewboard.asterisk.org/r/1563/
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