Per the bluez API, in order to bind to the first available port, the rc_channel
field of the socket addressing structure used to bind the socket should be set
to 0. Previously, Asterisk had set the rc_channel field set to 1, causing it
to connect to whatever happens to be on port 1.

We could probably not explicitly set rc_channel to 0 since we memset the struct
earlier, but explicitly setting it will hopefully prevent someone from coming
in and setting it to some explicit port in the future.

(closes issue ASTERISK-16357)
Reported by: challado
Tested by: Alexander Heinz, Nikolay Ilduganov, benjamin, eliafino, David van Geyn
patches:
  ASTERISK-16357.diff uploaded by Nikolay Ilduganov (license 6253)
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The description still claimed that it returned the number of messages rather than
whether there were messages waiting.
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'search' will look for any package containing the name provided, so we need to
force a more exact search.
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Since there is no need for the call-id logging ao2 object to have a lock,
don't create it with one.
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(issue ASTERISK-15456)
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Again, since res_jabber/res_xmpp have duplicate APIs, their documentation ref
links have to specify which reference they're referring to. The various
documentation parsers can interpret the module attribute however they want
in order to construct the appropriate links.
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........
  r379209 | mjordan | 2013-01-16 09:27:44 -0600 (Wed, 16 Jan 2013) | 8 lines
  
  Add module tags to documentation for res_jabber/res_xmpp
  
  Since res_jabber/res_xmpp provide the same APIs (app/func/manager/etc.),
  the XML documentation for each needs to call out which module is providing
  the documentation. The module attribute has been added to the various XML
  fragments for this purpose.
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  r379210 | mjordan | 2013-01-16 09:30:20 -0600 (Wed, 16 Jan 2013) | 4 lines
  
  Update the dtd to actually *support* the module attribute in all elements
  
  Mea culpa.
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The parser for SMS messages would incorrectly parse out the from number.
The parsing would incorrectly start scanning for the from number at the
same index as the first double quote ("); this would inadvertently cause
it to treat the first double quote as the terminating double quote for
the from number as well.

The SMSSRC should now populate correctly.

(closes issue ASTERISK-16822)
Reported by: menschentier
Tested by: Jonas Falck
patches:
 fixSMSSRC.patch uploaded by jonax (license 6320)

(closes issue ASTERISK-19153)
Reported by: Panos Gkikakis
patches:
  sms-sender-fix.diff uploaded by roeften (license 5884)  
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The chan_misdn channel driver will send a channel with an invalid destination
to the 'i' extension itself if said extension can be reached. It forgot,
however, to set the INVALID_EXTEN channel variable when it bounces the channel
to this extension. Dialplan writers everywhere moaned at yet another
inconsistency.

This is yet another example of why duplicating logic in multiple places results
in bugs that stick around in Jira for just under three years.

Yes: ASTERISK-15456 was created on January 18th, 2010. Patch committed on
January 15th, 2013. Ouch.

(closes issue ASTERISK-15456)
Reported by: Thomas Omerzu
patches:
  chan_misdn_invalid.patch2 uploaded by Thomas Omerzu (license 5927)
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From the patch author:

"First this patch adds general support for busy detection. It also adds support
 for the ECAM command at Sony Ericsson phones and also signals busy when only
 early media was received but the call got not answered."

Review: https://reviewboard.asterisk.org/r/323

(closes issue ASTERISK-14527)
Reported by: Artem Makhutov
Tested by: Artem Makhutov
patches:
  busy-full5.patch uploaded by artem (license 5757)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379144 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379128 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379070 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Masquerades are an insane implementation detail within Asterisk. It generates
a number of useless and confusing events, and manipulates channels in a way
that semantically doesn't make sense. I've given a fairly thorough review of
masquerade code and its usage on the wiki at
https://wiki.asterisk.org/wiki/x/IwBRAQ.

While ultimately it makes the most sense to abandon masquerades altogether,
it will take some time to completely irradicate. Even then, there may always
be code that's not worth rewriting to get rid of the masquerade.

This patch does two things to make masquerades slightly less insane:
 * When swapping the names of the original and clone channel, only emit a
   single rename event of original -> original<ZOMBIE>. The original code
   issued three rename events to accomplish the same end.
 * In addition to swapping the names of the channels, also swap their
   uniqueid's. This allows the 'Uniqueid' field to be used as a stable
   identifier for a channel from and external interface, such as AMI.

Review: https://reviewboard.asterisk.org/r/2266/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379023 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When r378933 was merged into 1.8, it should have also escaped
remote_display, since it will have the same XML encoding problem when
the caller/callee roles are reversed.

(closes issue ABE-2902)
Reported by: Guenther Kelleter
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In r370252 for ASTERISK-18404, Asterisk's handling of RTP was modified to
better account for out of order RTP packets. This was accomplished by using the
RTP timestamp and sequence number to check for out of order packets. However,
when a SSRC change occurs, the timestamp and sequence number will no longer
have any relation to the previously received packets. The variables tracking
the timestamp and sequence number therefore have to be reset.

(closes issue ASTERISK-20906)
Reported by: Eelco Brolman
patches:
  dtmf_on_hold.patch uploaded by Eelco Brolman (license #6442)
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XML encoding in chan_sip is accomplished by naively building the XML
directly from strings. While this usually works, it fails to take into
account escaping the reserved characters in XML.

This patch adds an 'ast_xml_escape' function, which works similarly to
'ast_uri_encode'. This is used to properly escape the local_display
attribute in XML formatted NOTIFY messages.

Several things to note:
 * The Right Thing(TM) to do would probably be to replace the
   ast_build_string stuff with building an ast_xml_doc. That's a much
   bigger change, and out of scope for the original ticket, so I
   refrained myself.
 * It is with great sadness that I wrote my own ast_xml_escape
   function. There's one in libxml2, but it's knee-deep in
   libxml2-ness, and not easily used to one-off escape a
   string.
 * I only escaped the string we know is causing problems
   (local_display). At least some of the other strings are
   URI-encoded, which should be XML safe. Rather than figuring out
   what's safe and escaping what's not, it would be much cleaner to
   simply build an ast_xml_doc for the messages and let the XML
   library do the XML escaping. Like I said, that's out of scope.

(closes issue ABE-2902)
Reported by: Guenther Kelleter
Tested by: Guenther Kelleter
Review: http://reviewboard.digium.internal/r/365/

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Previously if an XMPP client reconnected any filters added by an external module were lost.
This issue exhibited itself with chan_motif not receiving and reacting to Jingle signaling.

(closes issue ASTERISK-20916)
Reported by: kuj
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This provides a JSON API by pulling in and wrapping the Jansson JSON
library[1]. The Asterisk API basically mirrors the Jansson
functionality, with a few minor tweaks.

 * Some names have been asteriskified to protect the innocent.
 * Jansson provides both reference-stealing and reference-borrowing
   versions of several API's. The Asterisk API is exclusively
   reference-stealing for operations that put elements into arrays and
   objects.
 * No support for doubles, since we usually don't need that.
 * Coming along for the ride is the ast_test_validate macro, which made
   the unit tests much easier to write.

 [1]: http://www.digip.org/jansson/

(issue ASTERISK-20887)
(closes issue ASTERISK-20888)
Review: https://reviewboard.asterisk.org/r/2264/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378915 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fix an unbalanced manager_bridge_event(unlink) call if
AST_SOFTHANGUP_UNBRIDGE is set in ast_channel_bridge().

* Make ast_channel_bridge() use common cleanup code when leaving the
bridge.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378889 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Trivial changes in ast_channel_bridge().


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378874 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Squeezed some redundancy out of update_bridge_vars().

* Wrapped long line in __ast_change_name_nolink().


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378859 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* softmix_bridge_thread() was redundantly initializing an 8K buffer.

* Promoted a debug message to a warning in multiplexed_add_or_remove().


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* Made ast_test_init() match its prototype.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378790 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Minor optimization in ast_rtp_instance_early_bridge().


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378789 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The erroneous end condition would never include the AST_RTP_CISCO_DTMF flag
in the debug output.

(closes issue ASTERISK-20772)
Reported by: Xavier Hienne
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The prior location is before the declaration of struct ast_str, which causes
compiler warnings.

(closes issue ASTERISK-20852)
Reported by: Pavel Troller
Patches:
	strings.diff uploaded by Pavel Troller (license 6302)
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(closes issue ASTERISK-20826)
Reported by: snuffy
Patches:
	notabs.dif uploaded by snuffy (license 5024)
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(issue ASTERISK-16115)
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When ringinuse=no queue members can receive more than one call if these
calls happen at nearly the same time.

* Fix so a queue member does not receive more than one call from a queue.

NOTE: This fix does not prevent multiple calls to a member if the member
is in more than one queue.

* Did some refactoring to eliminate some code redundancy.

(issue ASTERISK-16115)
Reported by: nik600
Patches:
      jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett
      Modified

* Revert the -r341580 and -r341599 changes adding the queues.conf
check_state_unknown option as it was added in an attempt to fix this
problem.  The fix did not need to be optional.  The fix should not have
tried to explicitly set the device state.  Setting the device state by
something other than the device introduces a race condition.  I also could
not see how the change would be effective other than delaying the
app_queue code long enough for the device state to propagate to app_queue.
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Cleanup of red blobs in chan_skinny and possible other small formatting issues.

Review: https://reviewboard.asterisk.org/r/2262/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378634 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Above says it all. Code by snuff, cleaned up by me. 

Review: https://reviewboard.asterisk.org/r/2246/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378624 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This rewrite changes skinny dialing from the threaded simpleswitch
to a scheduled timeout approach. There were some underlying issues
with the threaded simple switch with occasional corruption and
possible segfaults.

Review: https://reviewboard.asterisk.org/r/2240/
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Under some circumstances, libsrtp's srtp_create function deallocates memory that
it wasn't initially responsible for allocating. Because we weren't initially
aware of this behavior, this memory was still used in spite of being unallocated
during the course of the srtp_unprotect function. A while back I made a patch
which would set this value to NULL, but that exposed a possible condition where
we would then try to check a member of the struct which would cause a segfault.
In order to address these problems, ast_srtp_unprotect will now set an error value
when it ends without a valid SRTP session which will result in the caller of
srtp_unprotect observing this error and hanging up the relevant channel instead of
trying to keep using the invalid session address.

(closes issue ASTERISK-20499)
Reported by: Tootai
Review: https://reviewboard.asterisk.org/r/2228/diff/#index_header
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On a fresh checkout of Asterisk 11, running make before ./configure
could cause the pjproject subdirectory to get in an odd state that
would prevent compilation. This patch by Tilghman prevents that from
occurring.

(closes issue ASTERISK-20681)
Reported by: Dinesh Ramjuttun
Tested by: danilo borges, Steve Lang
patches:
  20121208__ccar_solved.diff.txt uploaded by Tilghman Lesher (license 5003)
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On a multihomed server when sending a NOTIFY message, we were not figuring out
which network should be used to contact the peer.

This patch fixes the problem by calling ast_sip_ouraddrfor() and then
build_via() so that our NOTIFY message contains the correct IP address.

Also, a debug message is being added to help follow the call-id changes that
occur.  This was helpful for confirming that the IP address was set properly
since the call-id contains the IP address.  It also will be helpful for
troubleshooting purposes when following a call in the debug logs.

(closes issue ASTERISK-20805)
Reported by: Bryan Hunt
Tested by: Bryan Hunt, Michael L. Young
Patches:
    asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2255/
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(closes issue AST-1036)
Reported by: jbigelow
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378557 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Baseline clean up of formating to make room for extended documentation

(issue ASTERISK-20259)


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