res_http_websocket: Include query parameters in client connection requests.

Review: https://reviewboard.asterisk.org/r/3914/
........

Merged revisions 421210 from http://svn.asterisk.org/svn/asterisk/branches/13
configure: Undefine FORTIFY_SOURCE prior to defining it for patched gcc

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)
........

Merged revisions 421227 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 421228 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 421229 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421230 from http://svn.asterisk.org/svn/asterisk/branches/13
apps/app_dial: Fix Dial 'z' option

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)
........

Merged revisions 421232 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 421233 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 421234 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421235 from http://svn.asterisk.org/svn/asterisk/branches/13
apps/app_meetme: Fix crash when publishing MeetMe messages with no channel

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
........

Merged revisions 421270 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421273 from http://svn.asterisk.org/svn/asterisk/branches/13
Improve call forwarding reporting, especially with regards to ARI.

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
........

Merged revisions 420794 from http://svn.asterisk.org/svn/asterisk/branches/13
Multiple revisions 421311-421312

........
  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
........
  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.
........

Merged revisions 421311-421312 from http://svn.asterisk.org/svn/asterisk/branches/13
func_config: Change 'Not Found' message from ERROR to DEBUG

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/
........

Merged revisions 421327 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 421328 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 421329 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421337 from http://svn.asterisk.org/svn/asterisk/branches/13
Skinny: Fixup compile warning for non dev-mode.
........

Merged revisions 421376 from http://svn.asterisk.org/svn/asterisk/branches/13
chan_pjsip: Fix attended transfer connected line name update.

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/
........

Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13
ARI: Fix a bug where /channels/{channelID}/continue doesn't execute PBX

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)
........

Merged revisions 421416 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421423 from http://svn.asterisk.org/svn/asterisk/branches/13
AMI Docs: Fix Status channel parameter optionality
........

Merged revisions 421442 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 421443 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 421444 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421445 from http://svn.asterisk.org/svn/asterisk/branches/13
Fix compilation error on certain versions of GCC.
........

Merged revisions 421447 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421448 from http://svn.asterisk.org/svn/asterisk/branches/13
Alter documentation for callerid_privacy to use correct values.
........

Merged revisions 421485 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421488 from http://svn.asterisk.org/svn/asterisk/branches/13
AMI: Add AllVariables parameter to Status

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/
Stasis: Add information to blind transfer event

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/
........

Merged revisions 421537 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421538 from http://svn.asterisk.org/svn/asterisk/branches/13
Move evaluation of set_var options in pjsip to the end of channel initialization.

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.
........

Merged revisions 421565 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421566 from http://svn.asterisk.org/svn/asterisk/branches/13
Set the role for inbound subscriptions correctly.

This was causing the AMI show_subscriptions test in
the testsuite to fail since all subscriptions were being
seen as subscribers instead of notifiers.
........

Merged revisions 421585 from http://svn.asterisk.org/svn/asterisk/branches/13
cli.c: Fix tab completion of "module load" when MALLOC_DEBUG is enabled.

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().
........

Merged revisions 421600 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 421602 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 421608 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 421616 from http://svn.asterisk.org/svn/asterisk/branches/13
chan_pjsip: Update media translation paths when new SDP negotiated.

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/
........

Merged revisions 421645 from http://svn.asterisk.org/svn/asterisk/branches/13
uri: Quiet warning about type qualifiers ignored on function return type

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)
........

Merged revisions 421675 from http://svn.asterisk.org/svn/asterisk/branches/13