Properly detect launchd

Asterisk was a little too pro-active in claiming that it found launchd. On
systems without launchd - such as FreeBSD - this resulted in certain items
in Asterisk that conflict with launchd to not be selectable, such as
res_timing_kqueue.

(closes issue ASTERISK-20749)
Reported by: Oleg Baranov
........

Merged revisions 381847 from http://svn.asterisk.org/svn/asterisk/branches/1.8
app_dial: Honor the 'c' flag when the calling party hangs up

Apparently this feature became broken in 11, probably as a result
of the Hangup Cause project.

(closes issue ASTERISK-21113)
Reprted by: Heiko Wundram
Patches:
	app_dial.patch uploaded by Heiko Wundram (license 5822)
Fix FastAGI To Properly Check For A Connection

When IPv6 support was added to FastAGI, the intent was to have the ability to
check all addresses resolved for a host since we might receive an IPv4 address
and an IPv6 address.  The problem with the current code, is that, since we are
doing O_NONBLOCK, we get EINPROGRESS when calling ast_connect() but are ignoring
this instead of handling it.  We break out of the loop and continue on.  When we
later call ast_poll(), it succeeds but we never check if we have a connection or
not on the socket level.  We then attempt to send data to the host address that
we think is setup and it fails.  We then check the errno and see that we have
"connection refused" and then return with agi failed.

This patch does the following:

* Handles EINPROGRESS by creating the function handle_connection()
  - ast_poll() was moved into this function
  - This function checks the results of the connection on the socket level after
    calling ast_poll()
* Continues to the next address if the above fails to create a connection
* Once all addresses resolved are tried and we still are unable to establish a
  connection, then we return that the FastAGI call failed

(closes issue ASTERISK-21065)
Reported by: Jeremy Kister
Tested by: Jeremy Kister, Michael L. Young
Patches:
  asterisk-21065_poll_correctly_v4.diff Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2330/
Make ParkAndAnnounce return to priority + 1 when return context is not defined

The ParkAndAnnounce application documentation for the optional return_context
parameter states the following:

return_context
    The goto-style label to jump the call back into after timeout. Default
    'priority+1'.

Unfortunately, the application was sending the channel back into the dialplan
at 'priority', which is the ParkAndAnnounce application call. This causes an
infinite loop of the channel constantly being parked, announced, timed out,
parked, announced, timed out... while fun, especially for those callers you
wish to drive to the end of madness, this was not the intent of the
application.

(closes issue ASTERISK-20113)
Reported by: serginuez
patches:
  app_parkandannounce.diff uploaded by serginuez (License 6405)
........

Merged revisions 381916 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Don't display the AMI ALL class authorization for users if they don't have it

When converting AMI class authorizations to a string representation, the
method always appends the ALL class authorization. This is especially
important for events, as they should always communicate that class
authorization - even if the event itself does not specify ALL as a class
authorization for itself. (Events have always assumed that the ALL class
authorization is implied when they are raised)

Unfortunately, this did mean that specifying a user with restricted class
authorizations would show up in the 'manager show user' CLI command as
having the ALL class authorization.

Rather then modifying the existing string manipulation function, this patch
adds a function that will only return a string if the field being compared
explicitly matches class authorization field it is being compared against.
This prevents ALL from being returned unless it is actually specified for
the user.

(closes issue ASTERISK-20397)
Reported by: Johan Wilfer
........

Merged revisions 381939 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Set the sin_family on the bind address socket during initialization

Somehow, chan_jingle has managed to operate for years without setting the
sin_family on its bindaddr socket. This patch properly sets the field during
initial module load to AF_INET.

Note that the patch on the issue was modified slightly to change the
initialization of the socket from allocation of a chan_jingle private to the
module initialization, as the bindaddr object (which is static) only needs to
have the address set once.

(closes issue ASTERISK-19341)
Reported by: andre valentin
patches:
  0105-chan_jingle.patch uploaded by avalentin (License 6064)
........

Merged revisions 381975 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Clean up use of va_end/va_args in res_config_mysql

There were several problems using variadic argument macros in res_config_mysql.
 * Improper use of va_end. Multiple calls to va_end were possible resulting in
   an unbalanced matching of va_start/va_end.
 * Calls to va_arg after a possible encounter of a SENTINEL value.

This patch corrects those errors.

(closes issue ASTERISK-19451)
Reported by: wdoekes
patches:
  ASTERISK-19451-1.8--2.diff uploaded by wdoekes (License 5674)
........

Merged revisions 382021 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Ensure that the default bridge/user profiles are always available

ConfBridge and Page require that there always be a default bridge and user
profile available. While properties of the default profiles can be overriden
in the configuration file, removing them can create situations where neither
application can function properly.

This patch ensures that if an administrator removes the profiles from the
confbridge.conf configuration file, the profiles are added upon load.
Documentation clarifying this has been added to the confbridge.conf.sample file.

Review: https://reviewboard.asterisk.org/r/2356/

(closes issue AST-1115)
Reported by: John Bigelow
Tested by: John Bigelow
Clean up ConfBridge commands to account for wait_marked users

When ConfBridge was refactored to better handle the concept of marked,
wait_marked, and normal users co-existing in a conference (thereby implementing
a state machine for the conference), the wait_marked users were put into their
own list of conference participants, separate from the active users. This list
is used for wait_marked users when they are waiting in a conference but no
marked user has joined; normal users may have joined at this point however.
There are several AMI/CLI commands that affect conference users that were not
checking the wait_marked users list:
* CLI/AMI commands that mute/unmute a participant. In this case, wait_marked
  users have to remain in their particular state and should not be affected -
  however, the commands would return "Channel not found" as opposed to the
  appropriate error condition.
* CLI/AMI commands that kick a participant. An admin should always be able to
  kick a participant out of the conference.

This patch fixes both sets of commands, and cleans up the CLI commands slightly
by allowing them to complete a participant name (this was supposed to have been
added, but the function call was commented out and wasn't implemented).

Review: https://reviewboard.asterisk.org/r/2346/

(closes issue AST-1114)
Reported by: John Bigelow
Tested by: John Bigelow
Fix typo in r382068

Well, that was embarrassing. Removed an '-l' that somehow got in there.
Remove unneeded linux-gnueabi*

As of r380521 the configure scripts converts the value of linux-gnueabi*
of OSARCH to "linux-gnu". So no point in testing for those values.
........

Merged revisions 382087 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Correct RPID parsing for unquoted display-name.

Parsing Remote-Party-ID will now succeed if display-name is of the
*(token LWS) kind and not just the quoted-string kind.

Review: https://reviewboard.asterisk.org/r/2341/
........

Merged revisions 382107 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Consider linux-gnuspe as linux-gnu

* The powerpcspe Linux port uses linux-gnuspe as the OS string.
* Our build system shouldn't really care for that, so just call it linux-gnu.
* Original report: Roland Stigge , http://bugs.debian.org/701505

Review: https://reviewboard.asterisk.org/r/2357/
........

Merged revisions 382110 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Regenerate the configure script. The one in the tree was not working for me at all.
Relax dialog checking in get_sip_pvt_byid_locked so it works when the dialog is forked.

(closes issue ASTERISK-20638)
Reported by: eelcob
Patches:
      pedantic-call-pickup-from-tag.patch uploaded by eelcob (license 6442)
........

Merged revisions 382171 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Let channels joining a MeetMe conference opt out of the denoiser

For some channel drivers, specifically those that have a varying rate in the
number of audio samples, the audio quality for a MeetMe conference can be
exceedingly poor. This is due to a unilateral application of the DENOISE
function in func_speex to channels joining the conference.

The denoiser function in the speex library is initialized with the number of
audio samples in each sample that will be provided to it. If the number of
audio samples changes, the denoiser has to be thrown away and re-initialized.

While this could be worked around by removing func_speex, that doesn't help
if you actually use the denoiser with other channels on the system.

This patches does the following:
 * Checks for the presence of func_speex as opposed to codec_speex when
   determining if the DENOISE function is present (which is where the function
   is actually implemented)
 * Adds an option to MeetMe 'n' that causes the denoiser to not be applied
   to a channel when it joins. This keeps the current behavior the default, but
   let's users disable the denoiser if it causes problems on their system.

Review: https://reviewboard.asterisk.org/r/2358

(closes issue AST-1062)
Reported by: Thomas Arimont
........

Merged revisions 382227 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Prevent deadlock in chan_iax2 when attempting to set caller ID

A deadlock can occur in chan_iax2 when it attempts to set the caller ID, as it
already holds the iax2 private lock and improperly fails to obtain the channel
lock before calling ast_set_callerid. By not safely obtaining the channel lock,
a locking inversion can take place, causing a deadlock.

This patch solves this by calling the required deadlock avoidance functions
that obtain the channel lock before setting the caller ID.

Thanks to Pavel for fixing my syntax errors and testing this patch out.

(closes issue ASTERISK-21128)
Reported by: Pavel Troller
Tested by: Pavel Troller
patches:
  ASTERISK-21128-1.8.diff uploaded by mjordan (license 6283)
  ASTERISK-21128-modified-1.8.diff uploaded by Pavel Troller (license 6302)
........

Merged revisions 382233 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Fix a bug with ICE and strictrtp where media could get dropped.

If the end result of the ICE negotiation resulted in the path for media
changing it was possible for the strictrtp code to discard the RTP packets.
This change causes strictrtp to enter learning mode once again when the
ICE negotiation has completed successfully.
While the ICE negotiation is occurring leave strictrtp in an open state, media can and will come from different places.
Fix / Clean Up Some Items To Handle The New auto_* NAT Options

The original report had to do with a realtime peer behind NAT being pruned and
the peer's private address being used instead of its external address.  Upon
debugging, it was discovered that this was being caused by the addition of
the auto_force_rport and auto_comedia settings.

This patch does the following:

* Adds a missing note to the CHANGES file indicating that the default global nat
  setting is auto_force_rport

* Constify the 'req' parameter for check_via()

* Add calls to check_via() in a couple of places in order for the auto_*
  settings to do their job in attempting to determine if NAT is involved

* Set the flags SIP_NAT_FORCE_RPORT and SIP_PAGE2_SYMMETRICRTP if the auto_*
  settings are in use where it was needed

* Moves the copying of peer flags up in build_peer() to before they are used;
  this fixes the realtime prune issue

* Update the contrib/realtime schemas to allow the nat column to handle the
  different nat setting combinations we have

This patch received a review and "Ship It!" on the issue itself.

(closes issue ASTERISK-20904)
Reported by: JoshE
Tested by: JoshE, Michael L. Young
Patches:
  asterisk-20904-nat-auto-and-rt-peersv2.diff Michael L. Young (license 5026)