* Simplify compare_char() and avoid potential sign extension issue.

* Fix infinite loop in add_exten_to_pattern_tree() handling of character
set escape handling.

* Added buffer overflow checks in add_exten_to_pattern_tree() character
set collection.

* Made ignore empty character sets.

* Added escape character handling to end-of-range character in character
sets.  This has a slight change in behavior if the end-of-range character
is an escape character.  You must now escape it.

* Fix potential sign extension issue when expanding character set ranges.

* Made remove duplicated characters from character sets.  The duplicate
characters lower extension matching priority and prevent duplicate
extension detection.

* Fix escape character handling when the escape character is trying to
escape the end-of-string.  We could have continued processing characters
after the end of the exten string.  We could have added the previous
character to the pattern matching tree incorrectly.

(closes issue ASTERISK-18909)
Reported by: Luke-Jr
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The tlsenable settings are tucked away in main/tcptls.c, so I missed
them when resolving ASTERISK-18837. This should resolve the test suite
breakage of the sip tls tests.

Review: https://reviewboard.asterisk.org/r/1615
Reviewed by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347735 65c4cc65-6c06-0410-ace0-fbb531ad65f3

r325483 caused a regression in Asterisk 10+ that would make Asterisk segfault when
attempting to set penalty on an interface without specifying a queue in the queue set
penalty CLI command. In addition, no attempt would be made whatsoever to perform the
penalty setting on all the queues in the core list with either the cli command or the
non-segfaulting ami equivalent. This patch fixes that and also makes an attempt to
document and rename some functions required by this command to better represent what
they actually do. Oh yeah, and the use of this command without specifying a specific
queue actually works now.

Review: https://reviewboard.asterisk.org/r/1609/
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When a bridge is broken, ast_bridge_call() might execute the h exten on
the calling channel.  However, that channel may not have been the channel
that broke the bridge by hanging up.  The channel executing the h exten
must be in a hung up state so things like AGI run in the correct mode.

* Make sure ast_bridge_call() marks the channel it is executing the h
exten on as hung up.  (The AST_SOFTHANGUP_APPUNLOAD flag is used so as to
match the pbx.c main dialplan execution loop when it executes the h
exten.)

(closes issue ASTERISK-18811)
Reported by: David Hajek
Patches:
      jira_asterisk_18811_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: David Hajek, rmudgett
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AST-2011-014. When automon was enabled in features.conf, it was possible
to crash Asterisk by sending an INFO request if no channel had been
created yet.

(closes issue ASTERISK-18805)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347533 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix a segfault if an attempt to answer a call is made between when
the inbound call gives up (and the channel is removed) and when the
device is notified and removes the call from the device.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347490 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-18958)
Reported by: Red
Patches:
      jira_asterisk_18958_v1.8.patch (license #5621) patch uploaded by rmudgett
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Meetme would attempt to substitute the realtime values of RECORDING_FILE and
RECORDING_FORMAT from the meetme db entry instead of using the channel variable set
for those variables in spite of those database entries being NULL or even lacking
a column to represent them.

(closes issue ASTERISK-18873)
Reported by: Byron Clark
Patches:
	ASTERISK-18873-1.patch uploaded by Byron Clark (license 6157)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347395 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch also makes astdb2sqlite3 and astcanary use the configured
directory instead of relying on $PATH.

(closes issue ASTERISK-18959)
Review: https://reviewboard.asterisk.org/r/1613/
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(closes issue ASTERISK-18924)
Reported by: Kevin Taylor
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If waitExten specifies a music class to use with its music on hold option, it will use
CHANNEL(musicclass) instead if that channel variable has been set on the initiating
channel.  This documents that behavior in the waitExten app so that this can be known
without checking the documentation of the code in function local_ast_moh_start.

(closes issue ASTERISK-18804)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347241 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Deprecates MAILBOX_EXISTS. Provides count, email, exists, fullname,
language, locale, pager, password, tz.

(closes issue ASTERISK-18634)
Patch by: Kris Shaw
Review: https://reviewboard.asterisk.org/r/1568
Reviewed by: Walter Doekes


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347192 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When tcpenable=no, sending to transport=tcp hosts was still allowed.
Resolving the source address wasn't possible and yielded the string
"(null)" in SIP messages. Fixed that and a couple of not-so-correct
log messages.

(closes issue ASTERISK-18837)
Reported by: Andreas Topp

Review: https://reviewboard.asterisk.org/r/1585
Reviewed by: Matt Jordan
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Review: https://reviewboard.asterisk.org/r/1572
Reviewed by: Matt Jordan
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until after the mailboxes were initialized. This caused the settings to
be unset for those mailboxes until a reload was performed.

(closes issue ASTERISK-18838)

Review: https://reviewboard.asterisk.org/r/1570
Reviewed by: Matt Jordan
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Update the doubly linked list implementation.  Now safe traversing can
insert before and after the current node when traversing in either
direction.

Updated the linked lists unit test test_linkedlist to also test doubly
linked lists.  The old test_dlinkedlist requires a manual check of results
and probably should be removed.

Review: https://reviewboard.asterisk.org/r/1569/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347110 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch resolves the issue where MWI subscriptions are orphaned
by subsequent SIP SUBSCRIBE messages.  When a peer is removed, either
by pruning realtime SIP peers or by unloading / loading chan_sip, the
MWI subscriptions that were orphaned would still be on the event engine
list of valid subscriptions but have a pointer to a peer that no longer
was valid.  When an MWI event would occur, this would cause a seg fault.

(closes issue ASTERISK-18663)
Reported by: Ross Beer
Tested by: Ross Beer, Matt Jordan
Patches:
  blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283)

Review: https://reviewboard.asterisk.org/r/1610/
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Extracting sig_analog from chan_dahdi lost call progress detection
functionality.

* Fix analog ports from considering a call answered immediately after
dialing has completed if the callprogress option is enabled.

(closes issue ASTERISK-18841)
Reported by: Richard Miller
Patches:
      chan_dahdi.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
      sig_analog.c.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
      sig_analog.h.diff (license #5685) patch uploaded by Richard Miller
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Prior to this patch, if labels with the same name were used for different priorities in
the same extension, the new label would be accepted, but it would be unusable since
attempts to reach that label would just go to the first one. Now pbx.c detects this,
generates a warning in logs, and culls the label before adding it to the dialplan.

(closes issue ASTERISK-18807)
Reported by: Kenneth Shumard
Patches:
	pbx.c.patch uploaded by Kenneth Shumard (License 5077)
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Add missing symbol exports for ast_aji_client_destroy and ast_aji_buddy_destroy
for usage outside res_jabber.  Testing of these changes focused on res_jabber
itself, so this problem was missed.

Reported-by: Michael Spiceland
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The code that allowed admins to create users with domain-only uri's had
stopped to work in 1.8 because of the reqresp parser rewrites. This is
fixed now: if you have a [mydomain.com] sip user, you can register with
useraddr sip:mydomain.com. Note that in that case -- if you're using
domain ACLs (a configured domain list) -- mydomain.com must be in the
allow list as well.

Reviewboard r1606 shows a list of registration combinations and which
SIP response codes are returned.

Review: https://reviewboard.asterisk.org/r/1533/
Reviewed by: Terry Wilson

(closes issue ASTERISK-18389)
(closes issue ASTERISK-18741)
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The previous patch (r346040) incorrectly parsed the URI in the presence
of a port, e.g., user@hostname:port would fail as the port would be
double appended to the SIP message.  This patch uses the parse_uri function
to correctly parse the URI into its username and hostname parts, and places
them in the correct fields in the sip_pvt structure.

(issue ASTERISK-18903)
Review: https://reviewboard.asterisk.org/r/1597/
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https://origsvn.digium.com/svn/asterisk/branches/10

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  r346763 | may | 2011-12-02 20:42:32 +0400 (Fri, 02 Dec 2011) | 14 lines
  
  Merged revisions 346762 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r346762 | may | 2011-12-02 20:19:19 +0400 (Fri, 02 Dec 2011) | 7 lines
    
    process null frame pointer returned by ast_rtp_instance_read correctly
    
    (closes issue ASTERISK-16697)
    Reported by: under
    Patches: 
            segfault.diff (License #5871) patch uploaded by under
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346777 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The STUN socket must remain open between polls or the external address
seen by the STUN server is likely to change.  However, if the STUN request
poll fails then the STUN server address needs to be re-resolved and the
STUN socket needs to be closed and reopened.

* Re-resolve the STUN server address and create a new socket if the STUN
request poll fails.

* Fix ast_stun_request() return value consistency.

* Fix ast_stun_request() to check the received packet for expected message
type and transaction ID.

* Fix ast_stun_request() to read packets until timeout or an associated
response packet is found.  The stun_purge_socket() hack is no longer
required.

* Reduce ast_stun_request() error messages to debug output.

* No longer pass in the destination address to ast_stun_request() if the
socket is already bound or connected to the destination.

(closes issue ASTERISK-18327)
Reported by: Wolfram Joost
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/1595/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346709 65c4cc65-6c06-0410-ace0-fbb531ad65f3

183 is actually a session progress message.

(closes issue ASTERISK-18925)
Reported by: Sebastian Denz
Tested by: jrose
Patches:
	asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian Denz (License #6139)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346655 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346617 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Cleaning up chan_sip/tcptls file descriptor closing.

This patch attempts to eliminate various possible instances of undefined behavior caused
by invoking close/fclose in situations where fclose may have already been issued on a
tcptls_session_instance and/or closing file descriptors that don't have a valid index
for fd (-1). Thanks for more than a little help from wdoekes.

(closes issue ASTERISK-18700)
Reported by: Erik Wallin

(issue ASTERISK-18345)
Reported by: Stephane Cazelas

(issue ASTERISK-18342)
Reported by: Stephane Chazelas

Review: https://reviewboard.asterisk.org/r/1576/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346563 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch attempts to eliminate various possible instances of undefined behavior caused
by invoking close/fclose in situations where fclose may have already been issued on a
tcptls_session_instance and/or closing file descriptors that don't have a valid index
for fd (-1). Thanks for more than a little help from wdoekes.

(closes issue ASTERISK-18700)
Reported by: Erik Wallin

(issue ASTERISK-18345)
Reported by: Stephane Cazelas

(issue ASTERISK-18342)
Reported by: Stephane Chazelas

Review: https://reviewboard.asterisk.org/r/1576/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346525 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Update the documentation surrounding the use of MONITOR_EXEC to make it more clear
that it can be used for both Monitor() and MixMonitor() usage.

(closes issue ASTERISK-17413)
Reported by: David Woolley
Patches:
     issue18817_mixmonitor_queues_doc.diff by Michael L. Young (License #5026)
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Review: https://reviewboard.asterisk.org/r/1599


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

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

Review: https://reviewboard.asterisk.org/r/1600/
Reviewed by: Walter Doekes
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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
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Review: https://reviewboard.asterisk.org/r/1594
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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
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