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This commit adds GoSub alternatives to connected line, redirecting, and CCSS
macro hooks so that macro can finally be deprecated.  This also adds
deprecation warnings for those features when used and in documentation.

Review: https://reviewboard.asterisk.org/r/1760/
(closes issue SWP-4256)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357013 65c4cc65-6c06-0410-ace0-fbb531ad65f3

chan_iax2 to pass in the correct types.

chan_iax2 is the only consumer for the various ast_netsock_* functions in trunk
at this point, so this feels like a safe change to make.


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send_diversion=no will prevent Diversion headers from being added to SIP
requests. This doesn't prevent Diversion from being added with dialplan
such as with SIPAddHeader.

(closes issue ASTERISK-16862)
Reported by: rsw686
Review: https://reviewboard.asterisk.org/r/1769/


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This patch makes sure amaflags, accountcode, and userfield get copied
to the bridge CDR when set during a bridge (like via a custom feature).

(closes issue ASTERISK-16990)
Review: https://reviewboard.asterisk.org/r/1721/
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(closes issue ASTERISK-19011)
Reported by: Walter Doekes
Patches:
	issueA19011_combine_read_and_write_locks_WORK_IN_PROGRESS.patch uploaded by Walter Doekes (license 5674)
review: https://reviewboard.asterisk.org/r/1719/
review: https://reviewboard.asterisk.org/r/1622/
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(issue ASTERISK-19298)
Reported by: Dmitry Melekhov
Patches:
        change_gk_on_reload-1.patch (License #5415)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356848 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In r354890, a memory leak in app_voicemail was fixed by properly disposing of
the allocated heard/deleted pointers.  However, there are situations,
particularly when no messages are found in a folder, where these pointers are
not allocated and not NULL.  In that case, an invalid free would be attempted,
which could crash app_voicemail.  As there are a number of code paths where
this could occur, this patch uses the number of messages detected in the folder
before it attempts to free the pointers.  This resolves the crash detected in
the Asterisk Test Suite's check_voicemail_nominal test.
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The SIP TCP/TLS worker threads were created joinable but noone could join
them if they died on their own.

* Fix the SIP TCP/TLS worker threads to not be created joinable.

* _sip_tcp_helper_thread() only needs one parameter since the pvt
parameter is only passed in as NULL and never used.

(closes issue ASTERISK-19203)
Reported by: Steve Davies

Review: https://reviewboard.asterisk.org/r/1714/
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The patch for ASTERISK-19253 included properly shutting down the libsrtp
library in the case of module unload.  Unfortunately, not all distributions
have the srtp_shutdown call.  As such, this patch removes calling
srtp_shutdown.
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Currently, when using res_srtp, once the SRTP policy has been added to the
current session the policy is locked into place.  Any attempt to replace an
existing policy, which would be needed if the remote endpoint negotiated a new
cryptographic key, is instead rejected in res_srtp.  This happens in particular
in transfer scenarios, where the endpoint that Asterisk is communicating with
changes but uses the same RTP session.

This patch modifies res_srtp to allow remote and local policies to be reloaded
in the underlying SRTP library.  From the perspective of users of the SRTP API,
the only change is that the adding of remote and local policies are now added
in a single method call, whereas they previously were added separately.  This
was changed to account for the differences in handling remote and local
policies in libsrtp.

Review: https://reviewboard.asterisk.org/r/1741/

(closes issue ASTERISK-19253)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Patches:
  srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283)
  (with some small modifications for this check-in)
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Review: https://reviewboard.asterisk.org/r/1770/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Custom parking extensions may not be coded such that the first and only
extension priority is the Park application.  These custom parking
extensions will not be recognized as parking extensions.  When a call is
blind transferred to an extension that is not recognized as a parking
extension, the normal blind transfer code causes the transferred channel
to start executing dialplan.  Calls that get parked in this manner do not
know the original channel name that parked the call so the original parker
could never be called back if the parked call is not retrieved before the
timeout time.  The parking space is also announced to the call being
parked as a side effect of not knowing the original parking channel.

* Fix handling of BLINDTRANSFER channel variable for call parking.

* Fixed SIP blind transfer using the wrong dialplan context variable to
check for the parking extension.

(closes issue ASTERISK-19322)
Reported by: aragon
Tested by: rmudgett, jparker

Review: https://reviewboard.asterisk.org/r/1730/

JIRA AST-766
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When we send an ACK for a 2xx response to an INVITE, we are supposed
to use the learned route set. However, when we receive a non-2xx final
response to an INVITE, we are supposed to send the ACK to the same place
we initially sent the INVITE.

We had been doing this up until the changes went in that would build a route
set from provisional responses. That introduced a regression where we would
use the learned route set under all circumstances.

With this change, we now will set the destination of our ACK based on the
invitestate. If it is INV_COMPLETED then that means that we have received
a non-2xx final response (INV_TERMINATED indicates a 2xx response was received).
If it is INV_CANCELLED, then that means the call is being canceled, which
means that we should be ACKing a 487 response.

The other change introduced here is setting the invitestate to INV_CONFIRMED
when we send an ACK *after* the reqprep instead of before. This way, we can
tell in reqprep more easily what the invitestate is prior to sending the ACK.

(closes issue ASTERISK-19389)
reported by Karsten Wemheuer
patches:
    ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049)
	(with some slight modifications prior to commit)
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Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)
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  r356290 | pabelanger | 2012-02-22 15:20:29 -0500 (Wed, 22 Feb 2012) | 4 lines
  
  Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)
  
  Review: https://reviewboard.asterisk.org/r/1763/
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  r356335 | pabelanger | 2012-02-22 16:29:25 -0500 (Wed, 22 Feb 2012) | 2 lines
  
  Add back strsep() function for previous commit
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  r356337 | pabelanger | 2012-02-22 16:36:37 -0500 (Wed, 22 Feb 2012) | 2 lines
  
  Missed one strsep() function
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Review: https://reviewboard.asterisk.org/r/1766/


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If the res_calendar module was followed immediately by one of the
calendar tech modules and "core stop gracefully" was run, Asterisk
would crash.

This patch adds use count tracking for res_calendar so that it is
unloaded after the tech modules when shutting down gracefully. It
is now not possible to unload all the of the calendar modules via
"module unload res_calednar.so", but it is still possible to unload
them all via "module unload -h res_calendar.so".

Review: https://reviewboard.asterisk.org/r/1752/
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It now compiles, but there are some unrelated warnings for set but
unused variables.


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https://origsvn.digium.com/svn/asterisk/branches/10

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  r356215 | mjordan | 2012-02-22 08:53:53 -0600 (Wed, 22 Feb 2012) | 32 lines
  
  Merged revisions 356214 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r356214 | mjordan | 2012-02-22 08:50:20 -0600 (Wed, 22 Feb 2012) | 27 lines
    
    Fix potential buffer overrun and memory leak when executing "sip show peers"
    
    The "sip show peers" command uses a fix sized array to sort the current peers
    in the peers ao2_container.  The size of the array is based on the current
    number of peers in the container.  However, once the size of the array is
    determined, the number of peers in the container can change, as the peers
    container is not locked.  This could cause a buffer overrun when populating
    the array, if peers were added to the container after the array was created.
    Additionally, a memory leak of the allocated array would occur if a user
    caused the _show_peers method to return CLI_SHOWUSAGE.
    
    We now create a snapshot of the current peers using an ao2_callback with the
    OBJ_MULTIPLE flag.  This size of the array is set to the number of peers
    that the iterator will iterate over; hence, if peers are added or removed
    from the peers container it will not affect the execution of the "sip show
    peers" command.
    
    Review: https://reviewboard.asterisk.org/r/1738/
    
    (closes issue ASTERISK-19231)
    (closes issue ASTERISK-19361)
    Reported by: Thomas Arimont, Jamuel Starkey
    Tested by: Thomas Arimont, Jamuel Starkey
    Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan (license 6283)
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The accessors names for the "emulate_dtmf_digit" field on the ast_channel
are misleading. Change them to ast_channel_dtmf_digit_to_emulate*.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356183 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-19419)
PseudoReview - seanbright on IRC


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Move along, nothing to see here...
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Review: https://reviewboard.asterisk.org/r/1753/


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(closes issue ASTERISK-17176)
Reported by: John Covert
Tested by: Sean Bright
Patches:
   chan_iax2.c.qualifyfreqnotok.patch uploaded by John Covert (license 5512)
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Without this, the call to ast_sockaddr_set_port a few lines later is a noop.
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'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag.
sig_pri and sig_ss7 were keeping 'outgoing' flag insync.

Now provides a callback for all the low level sig_XXX modules.

(issue ASTERISK-19316)

alecdavis (license 585)
Reported by: Jeremy Pepper
Tested by: alecdavis
 
Review: https://reviewboard.asterisk.org/r/1747/
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