When issuing a POST /channels/{channel_id}/play on a channel that is not
yet answered, ARI is supposed to:
* Queue up an AST_CONTROL_PROGRESS on the channel
* Start up the playback of the media

Instead, we sneak an answer on the channel right before starting playing media.

This is due to ARI's usage of control_streamfile. This function implicitly
answers the channel (and doesn't give ARI the option to stop it). The answering
of the channel here is probably unnecessary:
* app_voicemail, by far the biggest consumer of this function, always answers
  the channels anyway
* control stream file (in res_agi) and ControlPlayback probably shouldn't be
  implicitly answering the channel. Answering should not be tied directly to
  playing back media.

As it turns out, the answering of the channel here is pretty old:
356042    twilson       if (ast_channel_state(chan) != AST_STATE_UP) {
  3087      anthm               res = ast_answer(chan);
180259   tilghman       }

(As in, ancient?)

Note that others ran into this problem and commented about it on various
mailing lists.

Review: https://reviewboard.asterisk.org/r/3907/

ASTERISK-24229 #close
Reported by: Matt Jordan
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Trivial patch to add new lines to several files missing them. This fixes
warnings when compiling with gcc 4.1.2 on CentOS 5.

ASTERISK-24245 #close
Reported by: Shaun Ruffell
patches:
  0002-Trivial-addition-of-newlines-at-end-of-three-files.patch uploaded by Shaun Ruffell (License 5417)
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This patch fixes gcc warnings that occur due to the type qualifier 'const'
being ignored on a return type of int.

ASTERISK-24246 #close
Reported by: Shaun Ruffell
patches:
  0001-main-uri-Quiet-warning-about-ignored-attribute-on-re.patch uploaded by Shaun Ruffell (License 5417)
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On a SIP reinvite that changes media strams, the PJSIP channel driver was
flooding the log with "Asked to transmit frame type %s, while native
formats is %s" warnings.

* Fixes PJSIP not setting up translation paths when the formats change on
a reinvite.  AFS-63 was effectively reintroduced because of the media
formats work.  res_pjsip_sdp_rtp.c:set_caps()

* Improved the unexpected frame format WARNING message to include more
information.

* Added protective locking while altering formats on a channel.  Reworked
set_format() to simplify and protect the formats under manipulation.

* Restored some code that got lost in the media_formats work.
(channel.c:set_format() and res_pjsip_sdp_rtp.c:set_caps())

AFS-137 #close
Reported by: Mark Michelson

Review: https://reviewboard.asterisk.org/r/3906/
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filename_completion_function() returns memory that was not allocated by
the MALLOC_DEBUG allocation tracker so the memory must be freed by
ast_std_free().
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This was causing the AMI show_subscriptions test in
the testsuite to fail since all subscriptions were being
seen as subscribers instead of notifiers.
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This allows for set_var to override certain defaults such as caller ID and codec
values. This also fixes a test suite regression. The "set_var" test suite test attempted
to use set_var to override caller ID, but a recent change caused that to no longer work.
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When a blind transfer occurs that is forced to create a local channel
pair to satisfy the transfer request, information about the local
channel pair is not published. This adds a field to describe that
channel to the blind transfer message struct so that this information
is conveyed properly to consumers of the blind transfer message.

This also fixes a bug in which Stasis() was unable to properly identify
the channel that was replacing an existing Stasis-controlled channel
due to a blind transfer.

Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3921/
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This adds the AllVariables parameter to the Status AMI action such that
if defined and set to "true", all channel variables will be reported in
the subsequent Status event(s). This parameter does not negate the
functionality of the "Variables" parameter so that global variables and
dialplan functions can be requested.

Review: https://reviewboard.asterisk.org/r/3915/


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If /channels/{channelID}/continue is called on a channel that was originated
without a PBX (such as the ARI command POST channel with a stasis application
argument), the channel will not start dialplan execution. This patch will now
run the PBX out of the stasis execution if the channel doesn't currently have
an active PBX upon continuing.

ASTERISK-24043 #close
Reported by: Krandon Bruse
Review: https://reviewboard.asterisk.org/r/3917/
Patches:
    stasis-continue.diff submitted by Krandon Bruse (license 6631)
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A calls B
B answers
B SIP attended transfers to C
C answers, B and C can see each other's connected line information
B completes the transfer
A has number but no name connected line information about C
  while C has the full information about A

I examined the incoming and outgoing party id information handling of
chan_pjsip and found several issues:

* Fixed ast_sip_session_create_outgoing() not setting up the configured
endpoint id as the new channel's caller id.  This is why party A got
default connected line information.

* Made update_initial_connected_line() use the channel's CALLERID(id)
information.  The core, app_dial, or predial routine may have filled in or
changed the endpoint caller id information.

* Fixed chan_pjsip_new() not setting the full party id information
available on the caller id and ANI party id.  This includes the configured
callerid_tag string and other party id fields.

* Fixed accessing channel party id information without the channel lock
held.

* Fixed using the effective connected line id without doing a deep copy
outside of holding the channel lock.  Shallow copy string pointers can
become stale if the channel lock is not held.

* Made queue_connected_line_update() also update the channel's
CALLERID(id) information.  Moving the channel to another bridge would need
the information there for the new bridge peer.

* Fixed off nominal memory leak in update_incoming_connected_line().

* Added pjsip.conf callerid_tag string to party id information from
enabled trust_inbound endpoint in caller_id_incoming_request().

AFS-98 #close
Reported by: Mark Michelson

Review: https://reviewboard.asterisk.org/r/3913/
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When you call the CONFIG dialplan function with the name of a variable that
doesn't exist in the target context you get an ERROR.  This does nothing but
clutter up the logs with messages that may be perfectly acceptable.  Just
because a variable wasn't in the context doesn't mean it's an error.  Maybei
t's optional or just needs to be defaulted or ignored.

This patch changes the log level from ERROR to DEBUG.  If a dialplan developer
wants to debug their dialplan they still canby setting the console debug level 
as needed.

Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3919/
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  r421311 | mjordan | 2014-08-17 20:11:28 -0500 (Sun, 17 Aug 2014) | 9 lines
  
  res/ari/resource_channels: Don't return allocation failure on failed function
  
  If a function fails to execute, it is most likely due to one of two reasons:
  (1) The function doesn't exist or can't be read from
  (2) The function is dangerous and is restricted based on the user's permissions
  
  Currently we return allocation failure, which is incorrect. This updates the
  reason code to more accurately reflect why the request failed.

  ASTERISK-24215
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  r421312 | mjordan | 2014-08-17 20:13:41 -0500 (Sun, 17 Aug 2014) | 4 lines
  
  res/ari/resource_channels: Fix compilation issue
  
  Forgot a parameter. Whoops.
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This patch addresses a few issues:

1) The order of Dial events have been changed when performing a call forward.
   The order has now been altered to
    1) Dial begins dialing channel A.
    2) When A forwards the call to B, we issue the dial end event to channel
       A, indicating the dial is being canceled due to a forward to B.
    3) When the call to channel B occurs, we then issue a new dial begin to
       channel B.

2) Call forwards are now reported on the calling channel, not the peer channel.

3) AMI DialEnd events have been altered to display the extension the call is
   being forwarded to when relevant.

4) You can now get the values of channel variables for channels that are not
   currently in the Stasis application. This brings the retrieval of channel
   variables more in line with the rest of channel read operations since they
   may be performed on channels not in Stasis.

ASTERISK-24134 #close
Reported by Matt Jordan

ASTERISK-24138 #close
Reported by Matt Jordan

Patches:
	forward-shenanigans.diff uploaded by Matt Jordan (License #6283)

Review: https://reviewboard.asterisk.org/r/3899
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The same function, meetme_stasis_generate_msg, handles creating and publishing
Stasis message both when there are channels in the MeetMe conference and when
there are no channels in the conference. When the performance improvement was
made to use cached snapshots, this created a situation where Asterisk would
crash: obtaining a cached snapshot is not NULL tolerant.

This patch restores the previous implementation, which used a NULL safe set
of routines to produce a blob containing the channel snapshot (if available)
and information about the MeetMe conference.

ASTERISK-24234 #close
Reported by: Shaun Ruffell
Tested by: Shaun Ruffell
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The 'z' option is supposed to disable the dial timeout in the case of a call
forward. Unfortunately, the wrong timeout timer was passed to the do_forward
function, resulting in the option not working.

ASTERISK-24225 #close
Reported by: dimitripietro
Tested by: dimitripietro
patches:
  jira_asterisk_24225_v1.8.patch uploaded by rmudgett (License 5621)
  jira_asterisk_24225_v11.patch uploaded by rmudgett (License 5621)
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Some distributions of Linux patch gcc to define FORTIFY_SOURCE when gcc is
executed with optimization. This "help" unfortunately results in re-definition
warnings when FORTIFY_SOURCE is later defined in Asterisk's build system. This
patch undefines FORTIFY_SOURCE prior to defining it to prevent this warning.

Review: https://reviewboard.asterisk.org/r/3912/

ASTERISK-24032 #close
Reported by: Kilburn
Tested by: Kilburn, wdoekes
patches:
  1.8.diff uploaded by cloos (License 5956)
  10.diff uploaded by cloos (License 5956)
  11.diff uploaded by cloos (License 5956)
  12.diff uploaded by cloos (License 5956)
  13.diff uploaded by cloos (License 5956)
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Review: https://reviewboard.asterisk.org/r/3914/
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r420934 introduced some failures in the test suite.  Upon investigating, it was
discovered that differences in the way we were evaluating whether a channel was in
the process of leaving a bridge were causing some reinvites not to occur (mostly
reinvites back to Asterisk when ending a call). This patch fixes that behavioral
change.

ASTERISK-24027 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3910/
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Moving the test event raised when a file is played back (which occurred in
r421059) broke the ever loving snot out of the voicemail tests. This caused
duplicate test events to get raised, as app_voicemail and main/app were raising
events prior to call ast_streamfile. The voicemail tests did not enjoy getting
multiple events.

Since raising the playback event in ast_streamfile is far more useful to the
vast majority of tests, this patch keeps the call there and simply removes the
extraneous calls that duplicated the event.
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The res_hep_rtcp module was incorrectly including <pjsip.h>. This didn't need
to be included, as the module does not using PJPROJECT any fashion.
Unfortunately, because res_hep_rtcp did not include pjsip in its MODULEINFO as
a dependency, this also meant that res_hep_rtcp will fail to compile on a
system without PJPROJECT.

This patch removes the include.

Thanks to Damien Wedhorn for pointing this out in #asterisk-dev.

ASTERISK-24236 #close
Reported by: Damien Wedhorn, Matt Jordan
Tested by: Damien Wedhorn
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This is being done in advance of the test for ASTERISK-23953
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CEL typically tracks a lot of information using the unique ID of the channel.
This is typically needed due to tying events together using the linked ID of
the various channels involved in a "call", which is derived from the channel ID
of the oldest channel involved in a bridge (or in the case of a Dial, the
parent channel).

Previously, we had updated the extra fields to include the involved channel
names, but forgot to put in the unique ID. This patch corrects that error.
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Reduce the scope of local_peer and only get it if the ARI originate is
subscribing to the channels.

Review: https://reviewboard.asterisk.org/r/3905/
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Use ao2_replace() instead of ao2_cleanup(); ao2_bump().

ao2_replace() has the advantange of not altering the ref count if the
replaced pointer is the same.

Review: https://reviewboard.asterisk.org/r/3904/
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This prevents a crash from occurring when a contact with no URI is used
for the creation of an outbound out-of-dialog request with no
associated endpoint.
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If a manager or CLI user attached a mixmonitor to a call running a dynamic
bridge feature while in a bridge, the feature would be interrupted and the
channel would be forcibly kicked out of the bridge (usually ending the call
during a simple 1 to 1 call). This would also occur during any similar action
that could set the unbridge soft hangup flag, so the fix for this was to
remove unbridge from the soft hangup flags and make it a separate thing all
together.

ASTERISK-24027 #close
Reported by: mjordan
Review: https://reviewboard.asterisk.org/r/3900/
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In r399267, the verbose2magic stuff was edited. This time it results
in magic characters in the log files for multiline messages.

In trunk (and 13) this was fixed by the "stripping" of those
characters from multiline messages (in r414798).

This fix is altered to actually strip the characters and not replace
them with blanks.

Review: https://reviewboard.asterisk.org/r/3901/
Review: https://reviewboard.asterisk.org/r/3902/
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Symptom is most likely an invalid ao2 object bad magic number message or a
less likely crash.
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* Made use ast_copy_string() instead of strcpy() for snoop uniqueid for
safety.  There is no guarantee that the max channel uniqueid length will
remain the same as the snoop uniqueid space.
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This is to support the backwards compatible changes made in the next version
of Asterisk.
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Return the correct value instead of always returning 0 when setting
internal status on unreal channels.

Reported by: Richard Mudgett
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