git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378823 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378783 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378748 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-20826)
Reported by: snuffy
Patches:
	notabs.dif uploaded by snuffy (license 5024)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378735 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-16115)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378691 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378688 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378623 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378593 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378585 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378565 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(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)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378543 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When the "h" extension is present within the context of the queue, all calls
are being reported COMPLETECALLER even when the agent is hanging up the call.

This patch checks to see if the agent hung-up or not instead of only relying on
checking if the queue (caller) channel hung-up or not.  It would appear that
having the h extension in the mix, the pbx goes to the h extension,
"hanging-up" the queue channel and triggering the reporting of COMPLETECALLER.

(closes issue ASTERISK-20743)
Reported by: call
Tested by: call, Michael L. Young
Patches: 
    asterisk-20743-q-cmplt-caller.diff 
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2256/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378516 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made agent_cont_sleep() and agent_ack_sleep() stop waiting if the wrapup 
time expires.  agent_cont_sleep() had tried but returned the wrong value 
to stop waiting.  

* Made agent_ack_sleep() take a struct agent_pvt pointer instead of a void
pointer for better type safety.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378488 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This test event was missing from channel.c causing the dial_LS_options
test to fail intermittently because of a race condition where most code
paths emitted the test event but this one did not. The dial_LS_options
test should stop bouncing now.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378460 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fix off-nominal path resource cleanup in agent_request().

* Create agent_pvt_destroy() to eliminate inlined versions in many places.

* Pull invariant code out of loop in add_agent().

* Remove redundant module user references in login_exec().

* Remove unused struct agent_pvt logincallerid[] member.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378458 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Avoid deadlock potential with local channels and simplify the locking.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378429 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is an interesting feature that allows additional strings to be used to
search the Directory, primarily intended to be used with nicknames, but could
be used with affiliations and the like.  Because the name field is used in
more than one place (such as email notifications), it is important that these
additional strings not be placed in the name field, but be specified
separately.

Review: https://reviewboard.asterisk.org/r/2244/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378414 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch changes res_xmpp to no longer cache events under certain circumstances.

(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378412 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Similar to r378287, res_xmpp was marshaling data read from an external source
onto the stack. For a sufficiently large message, this could cause a stack
overflow. This patch modifies res_xmpp in a similar fashion to res_jabber by
removing the stack allocation, as it was unnecessary.

(issue ASTERISK-20658)
Reported by: wdoekes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378410 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When parsing arguments, application entry points should not attempt to
directly modify the parameters to the function. This patch properly duplicates
the passed in parameters before attempting to parse them.

(issue ASTERISK-20658)
Reported by: wdoekes
patches:
  issueA20658_sanitize_app_mysql.patch uploaded by wdoekes (license 5674)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378384 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When reading configuration data from an Asterisk .conf file or when pulling
data from an Asterisk RealTime backend, Asterisk was copying the data on the
stack for manipulation. Unfortunately, it is possible to read configuration
data or realtime data from some data source that provides a large blob of
characters. This could potentially cause a crash via a stack overflow.

This patch prevents large sets of data from being read from an ARA backend or
from an Asterisk conf file.

(issue ASTERISK-20658)
Reported by: wdoekes
Tested by: wdoekes, mmichelson
patches:
 * issueA20658_dont_process_overlong_config_lines.patch uploaded by wdoekes (license 5674)
 * issueA20658_func_realtime_limit.patch uploaded by wdoekes (license 5674)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378377 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The AMI redirect action can fail to redirect two channels that are bridged
together.  There is a race between the AMI thread redirecting the two
channels and the bridge thread noticing that a channel is hungup from the
redirects.

* Made the bridge wait for both channels to be redirected before exiting.

* Made the AMI redirect check that all required headers are present before
proceeding with the redirection.

* Made the AMI redirect require that any supplied ExtraChannel exist
before proceeding.  Previously the code fell back to a single channel
redirect operation.

(closes issue ASTERISK-18975)
Reported by: Ben Klang

(closes issue ASTERISK-19948)
Reported by: Brent Dalgleish
Patches:
      jira_asterisk_19948_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Thomas Sevestre, Deepak Lohani, Kayode

Review: https://reviewboard.asterisk.org/r/2243/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378374 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Asterisk maintains an internal cache for devices in the event subsystem. The
device state cache holds the state of each device known to Asterisk, such that
consumers of device state information can query for the last known state for
a particular device, even if it is not part of an active call. The concept of
a device in Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls are allowed
in Asterisk. A device was automatically created and stored in the cache for
each anonymous call that occurred; this was possible in the SIP and IAX2
channel drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
are never removed from the system, allowing anonymous calls to potentially
exhaust a system's resources.

This patch changes the event cache subsystem and device state management to
no longer cache devices that are not associated with a physical entity.

(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
patches:
  event-cachability-3.diff uploaded by jcolp (license 5000)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378322 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Asterisk had several places where messages received over various network
transports may be copied in a single stack allocation. In the case of TCP,
since multiple packets in a stream may be concatenated together, this can
lead to large allocations that overflow the stack.

This patch modifies those portions of Asterisk using TCP to either
favor heap allocations or use an upper bound to ensure that the stack will not
overflow:
 * For SIP, the allocation now has an upper limit
 * For HTTP, the allocation is now a heap allocation instead of a stack
   allocation
 * For XMPP (in res_jabber), the allocation has been eliminated since it was
   unnecesary.

Note that the HTTP portion of this issue was independently found by Brandon
Edwards of Exodus Intelligence.

(issue ASTERISK-20658)
Reported by: wdoekes, Brandon Edwards
Tested by: mmichelson, wdoekes
patches:
  ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049)
  issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674)
  issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378288 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In ASTERISK-20726 UUID was added to Asterisk.  This commit is to add the dependancies to the install script



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378259 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378249 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378248 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This ensures that Asterisk rejects encrypted media streams (RTP/SAVP
audio and video) that are missing cryptographic keys and ensures that
the incoming SDP is consistent with RFC4568 as far as having a crypto
attribute present for any SAVP streams.

Review: https://reviewboard.asterisk.org/r/2204/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378166 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds hangup-related test events in order to support testing
of time-limited bridges. This aids in testing the S() and L() bridge
options.

(issue SWP-4713)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378095 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The chan_local module references were manually tied to the existence of
the ;1 and ;2 channel links.

* Made chan_local module references tied to the existence of the local_pvt
structure as well as automatically take care of the module references.

* Tweaked the wording of the local_fixup() failure warning message to make
sense.

Review: https://reviewboard.asterisk.org/r/2181/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378091 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fix local_alloc() unexpected limitation of exten and context length from
a combined length of 80 characters to a normal 80 characters each.

* Made local_alloc() and local_devicestate() parse the same way.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378081 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This was causing issues when using DESTDIR, since the path to which the link
pointed is not likely to exist (and not useful to exist) on the target system.

(issue ASTNOW-284)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378074 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* awesome_locking() does not need to thrash the pvt lock as much.

* local_setoption() does not need to check for NULL pvt on cleanup since
it will never be NULL.

* Made ref the pvt before locking for consistency.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378072 65c4cc65-6c06-0410-ace0-fbb531ad65f3