Allow specifying a port number for the MySQL server.

This patch allows you to specify a port number for the MySQL server.
It's useful if a MySQL server is running on a non-standard port.
Even though this module is deprecated in favor of func_odbc, someone
asked for this feature and it seems pretty harmless to add.

It has been tested using a number of combinations of with/without a
port number specified in the dialplan and changing the port number
 for mysqld.
Add a "corosync ping" CLI command.

This patch adds a new CLI command to the res_corosync module.  It is primarily
used as a debugging tool.  It lets you fire off an event which will cause
res_corosync on other nodes in the cluster to place messages into the logger if
everything is working ok.  It verifies that the corosync communication is
working as expected.

I didn't put anything in the CHANGES file for this, because this module is new
in Asterisk 11.  There is already a generic "res_corosync new module" entry in
there so I figure that covers it just fine.
Fix ast_event_new unit test.

One of my recent commits broke this test.  The error was:

[test_event.c:event_new_test:214]: Events expected to be identical
have different size: 69 != 59

The difference in size occurred because the first event had
the EID IE added to the event twice.  ast_event_new() now always
adds it automatically.  Previously it only added it if there
were no IEs specified, which was kind of weird.
app_meetme: Change app_meetme support level to extended from deprecated

(closes issue ASTERISK-20134)
Reported by: Leif Madsen
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Merged revisions 370547 from http://svn.asterisk.org/svn/asterisk/branches/10
Release B channel allocation on error path in chan_misdn.
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DECLINE to load confbridge if the config fails to load.
Fix some presence-state unit test typos.
Tweak unit test warning message.
Clean up chan_sip

This clean up was broken out from
https://reviewboard.asterisk.org/r/1976/ and addresses the following:
 - struct sip_refer converted to use the stringfields API.
 - sip_{refer|notify}_allocate -> sip_{notify|refer}_alloc to match
   other *alloc functions.
 - Replace get_msg_text, get_msg_text2 and get_pidf_body -> No, not
   get_pidf_msg_text_body3 but get_content, to match add_content.
 - get_body doesn't get the request body, renamed to get_content_line.
 - get_body_by_line doesn't get the body line, and is just a simple if
   test. Moved code inline and removed function.
 - Remove camelCase in struct sip_peer peer state variables,
   onHold -> onhold, inUse -> inuse, inRinging -> ringing.
 - Remove camelCase in struct sip_request rlPart1 -> rlpart1,
   rlPart2 -> rlpart2.
 - Rename instances of pvt->randdata to pvt->nonce because that is what
   it is, no need to update struct sip_pvt because _it already has a
   nonce field_.
 - Removed struct sip_pvt randdata stringfield.
 - Remove useless (and inconsistent) 'header' suffix on variables in
   handle_request_subscribe.
 - Use ast_strdupa on Event header in handle_request_subscribe to avoid
   overly complicated strncmp calls to find the event package.
 - Move get_destination check in handle_request_subscribe to avoid
   duplicate checking for packages that don't need it.
 - Move extension state callback management in handle_request_subscribe
   to avoid duplicate checking for packages that don't need it.
 - Remove duplicate append_date prototype.
 - Rename append_date -> add_date to match other add_xxx functions.
 - Added add_expires helper function, removed code that manually added
   expires header.
 - Remove _header suffix on add_diversion_header (no other header adding
   functions have this).
 - Don't pass req->debug to request handle_request_XXXXX handlers if req
   is also being passed.
 - Don't pass req->ignore to check_auth as req is already being passed.
 - Don't create a subscription in handle_request_subscribe if
   p->expiry == 0.
 - Don't walk of the back of referred_by_name when splitting string in
   get_refer_info
 - Remove duplicate check for no dialog in handle_incoming when
   sipmethod == SIP_REFER, handle_request_refer checks for that.

Review: https://reviewboard.asterisk.org/r/1993/
Patch-by: gareth
Add "dialplan remove context" and modify "dialplan add include"

From corruptor's review board posting:

"I've noticed that we can remove particular extension from context with
dialplan remove extension command but in order to remove all extensions
in the context we should delete them on by one. I've created dialplan
remove context command which uses ast_context_destroy to destroy the
whole context with all extensions. I've created to functions for in
pbx_config.c: handle_cli_dialplan_remove_context which actually removes
context and complete_dialplan_remove_context which completes input.
They are based on other similar functions and pretty trivial but I can be
mistaken somewhere.

"I've also modified dialplan add include <context2> into <context1>. I've
made it similar dialplan add extension ... command. It creates <context1>
if it doesn't exist and I've also modified complete_dialplan_add_include
and removed check for existance of <context2> because we can include
non-existent context into another one. (I usually include empty
(non-existent) contexts in advance). Should we raise warning in this case
as it's raised while reading extensions.conf?

"I use those functions with AMI. I think manager commands should be created
in addition to those CLI commands."

I've addressed the latest comments on review board and have made some other
coding guidelines-related cleanup. I also have modified the CHANGES file to
mention these new commands.

(closes issue ASTERISK-19292)
reported by Andrey Solovyev

Patches:
	dialplan_add_include.patch
    uploaded by Andrey Solovyev (license #5214)
    dialplan_remove_context.patch
    uploaded by Andrey Solovyev (license #5214)

Review: https://reviewboard.asterisk.org/r/2042
Clean up and ensure proper usage of alloca()

This replaces all calls to alloca() with ast_alloca() which calls gcc's
__builtin_alloca() to avoid BSD semantics and removes all NULL checks
on memory allocated via ast_alloca() and ast_strdupa().

(closes issue ASTERISK-20125)
Review: https://reviewboard.asterisk.org/r/2032/
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Move event cache updates into event processing thread.

Prior to this patch, updating the device state cache was done by the thread
that originated the event.  It would update the cache and then queue the event
up for another thread to dispatch.  This thread moves the cache updating part
to be in the same thread as event dispatching.

I was working with someone on a heavily loaded Asterisk system and while
reviewing backtraces of the system while it was having problems, I noticed that
there were a lot of threads contending for the lock on the event cache.  By
simply moving this into a single thread, this helped performance *a lot* and
alleviated some deadlock-like symptoms.

Review: https://reviewboard.asterisk.org/r/2066/
Schedule pokes of registered SIP peers within a given timespan after SIP reload

With a large number of SIP peers registered, performing a SIP reload causes a
flood of SIP OPTIONS request packets.  These are immediately sent out, and, as
responses come back, can cause peers to be flagged as 'lagged' due to handling
of the many response messages.

This fix prevents this "packet storm" and schedules the pokes for a random
time.  That time varies between 1 ms and the peer's qualify time, or, if
the qualify time is unknown, the global qualifyfreq setting.

The committed patch has some very small modifications to the patch schmidts
wrote for the review.

(closes issue ASTERISK-19154)
Reported by: Nicolo Mazzon
patches:
  issue19154.patch license #6034 uploaded by schmidts

Review: https://reviewboard.asterisk.org/r/1652
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Add "setvar" option to manager.conf.

With this option set, channel variables can be set on
every manager originate. The Variable header can still
be used to set additional channel variables for individual
calls if desired.

This work was completed by Olle Johansson on review board.
I have applied the review feedback and am committing it in
order to get this into trunk before Asterisk 11 is branched.

Review: https://reviewboard.asterisk.org/r/1412
Add headers from SIPAddHeader to outbound REFER requests.

This is a patch from kkm from review board.

This is useful for adding headers to REFER requests that
emanate from a Transfer() dialplan application call.

This also fixes some uses of the Referred-by header, removing
an extra set of angle brackets.

I've modified the reporter's original patch to not require
any additions to the sip_refer header and to just remove the
referred_by_name from sip_refer since it is no longer needed
or used.

(closes Issue ASTERISK-17639)
reported by Kirill Katsnelson
Patches:
	019059-sip-refer-addheaders-trunk-353549.diff
	uploaded by Kirill Katsnelson (license #5845)

Review: https://reviewboard.asterisk.org/r/1159
Revert alloca changes for utils

These changes were a tad overzealous in the utils directory.
Unfortunately, these don't compile with a "make".
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Make astobj2.h not include linkedlists.h.

Using astobj2 does not require linkedlists.h be included even though
astob2 uses linked lists internally.
Fix a possible crash due to passing NULL to ast_variables_dup()
Fix regression from r370636

When the chan_sip cleanup went in, a typo was included that caused some
subscriptions of non-Polycom phones to be limited to the same
capabilities as Polycom phones. This resolves the failures in the test
suite resulting from this regression.
Multiple revisions 370769-370771

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  r370769 | mmichelson | 2012-08-03 16:35:00 -0500 (Fri, 03 Aug 2012) | 24 lines
  
  Fix error in the "IPorHost" section of a SIP dialstring.
  
  This is based on the review request posted by Walter Doekes
  (referenced lower in the commit message)
  
  The main fix here is to treat the IPorHost portion of the dial
  string as a temporary outbound proxy. This ensures requests
  get sent to the proper location.
  
  Due to the age of the request, some parts were no longer relevant.
  For instance, the request moved outbound proxy parsing code into
  a single method. This is done in a previous commit, so it was not
  necessary to do again.
  
  Also, the review request fixed some errors with regards to request
  routing for CANCEL and ACK requests. This has also been fixed in
  more recent commits.
  
  (closes issue ASTERISK-19677)
  reported by Walter Doekes
  
  Review https://reviewboard.asterisk.org/r/1859
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  r370770 | mmichelson | 2012-08-03 16:39:35 -0500 (Fri, 03 Aug 2012) | 3 lines
  
  Remove unused variable.
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  r370771 | mmichelson | 2012-08-03 16:43:52 -0500 (Fri, 03 Aug 2012) | 5 lines
  
  Seriously? Another compilation error fixed.
  
  Somebody beat me.
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