res_rtp_asterisk: Add support for DTLS handshake retransmissions

On congested networks, it is possible for the DTLS handshake messages to get
lost. This patch adds a timer to res_rtp_asterisk that will periodically
check to see if the handshake has succeeded. If not, it will retransmit the
DTLS handshake.

Review: https://reviewboard.asterisk.org/r/3337

ASTERISK-23649 #close
Reported by: Nitesh Bansal
patches:
  dtls_retransmission.patch uploaded by Nitesh Bansal (License 6418)


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413008 f38db490-d61c-443f-a65b-d21fe96a405b
Websocket: Add session locking and delay close

This resolves a race condition where data could be written to a NULL
FILE pointer causing a crash as a websocket connection was in the
process of shutting down by adding locking to websocket session writes
and by deferring session teardown until session destruction.

(closes issue ASTERISK-23605)
Review: https://reviewboard.asterisk.org/r/3481/
Reported by: Matt Jordan


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413123 f38db490-d61c-443f-a65b-d21fe96a405b
chan_sip.c: Fixed off-nominal message iterator ref count and alloc fail issues.

* Fixed early exit in sip_msg_send() not destroying the message iterator.

* Made ast_msg_var_iterator_next() and ast_msg_var_iterator_destroy()
tolerant of a NULL iter parameter in case ast_msg_var_iterator_init()
fails.

* Made ast_msg_var_iterator_destroy() clean up any current message data
ref.

* Made struct ast_msg_var_iterator, ast_msg_var_iterator_init(),
ast_msg_var_iterator_next(), ast_msg_var_unref_current(), and
ast_msg_var_iterator_destroy() use iter instead of i.


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413139 f38db490-d61c-443f-a65b-d21fe96a405b
Return the number of rows affected by a SQL insert, rather than an object ID.

The realtime API specifies that the store callback is supposed to return the number
of rows affected. res_config_pgsql was instead returning an Oid cast as an int, which
during any nominal execution would be cast to 0. Returning 0 when more than 0 rows were
inserted causes problems to the function's callers.

To give an idea of how strange code can be, this is the necessary code change to fix
a device state issue reported against chan_pjsip in Asterisk 12+. The issue was that
the registrar would attempt to insert contacts into the database. Because of the 0
return from res_config_pgsql, the registrar would think that the contact was not successfully
inserted, even though it actually was. As such, even though the contact was query-able
and it was possible to call the endpoint, Asterisk would "think" the endpoint was unregistered,
meaning it would report the device state as UNAVAILABLE instead of NOT_INUSE.

The necessary fix applies to all versions of Asterisk, so even though the bug reported
only applies to Asterisk 12+, the code correction is being inserted into 1.8+.

Closes issue ASTERISK-23707
Reported by Mark Michelson
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Merged revisions 413224 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413225 f38db490-d61c-443f-a65b-d21fe96a405b
Prevent crashes in res_config_odbc due to uninitialized string fields.

Patches:
    odbc-crash.patch by John Hardin (License #6512)
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Merged revisions 413241 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413251 f38db490-d61c-443f-a65b-d21fe96a405b
Ensure that all parts of SQL UPDATEs and DELETEs are encoded.

Patches:
	res_config_odbc.patch by John Hardin (License #6512)
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Merged revisions 413304 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413305 f38db490-d61c-443f-a65b-d21fe96a405b
Fix encoding of custom prepare extra data.

Patches:
	res_config_odbc-take2.patch by John Hardin (License #6512)
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Merged revisions 413396 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413397 f38db490-d61c-443f-a65b-d21fe96a405b
app_confbridge: Fix ref leak in CLI "confbridge kick" command.

Fixed ref leak in the CLI "confbridge kick" command when the channel to be
kicked was not in the conference.


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413451 f38db490-d61c-443f-a65b-d21fe96a405b
app_queue: Extend documentation for various Manager actions and events.
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Merged revisions 413485 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413486 f38db490-d61c-443f-a65b-d21fe96a405b
app_chanspy: Fix a bug where Barge mode could fail

If the barge audiohook was attached prior to the spyee and its peer
actually being bridged, the audiohook would not be applied and the
connected peer would not be able to hear audio from the spy when the
spy is in barge mode.

(closes issue ASTERISK-23381)
Reported by: Robert Moss
Review: https://reviewboard.asterisk.org/r/3505/


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413551 f38db490-d61c-443f-a65b-d21fe96a405b
Allow Asterisk to compile under GCC 4.10

This resolves a large number of compiler warnings from GCC 4.10 which
cause the build to fail under dev mode. The vast majority are
signed/unsigned mismatches in printf-style format strings.
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Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413587 f38db490-d61c-443f-a65b-d21fe96a405b
Fix 32bit build for func_env
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Merged revisions 413592 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413595 f38db490-d61c-443f-a65b-d21fe96a405b
Blocked revisions 413591

........
Fix 32bit build for chan_sip


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413695 f38db490-d61c-443f-a65b-d21fe96a405b
app_chanspy: Fix a test that was failing on account of r413551

ASTERISK-23381 #close
ASTERISK-23381 #comment Reported by: Robert Moss
Review: https://reviewboard.asterisk.org/r/3505/


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413710 f38db490-d61c-443f-a65b-d21fe96a405b
chan_dahdi/sig_pri: Prevent unnecessary PROGRESS events when overlap dialing is enabled.

When overlap dialing is enabled, the lack of inband audio available
information in the SETUP_ACKNOWLEDGE events causes an interoperability
problem with SIP.  sig_pri doesn't know if there is dialtone present when
a SETUP_ACKNOWLEDGE is received so it assumes it is there and posts an
AST_CONTROL_PROGRESS frame.  The SIP channel driver then sends out a 183
Session Progress and blocks the desired 180 Ringing message when the
ALERTING message comes in.

* Made the configure script detect if the installed version of libpri
supports the SETUP_ACKNOWLEDGE enhancements.

* Using the new API, made generate an AST_CONTROL_PROGRESS frame on an
incoming SETUP_ACKNOWLEDGE message when the message indicates inband audio
is present instead of assuming that dialtone is present.

* Using the new API, made SETUP_ACKNOWLEDGE send out an inband audio
available indication only if dialtone is expected.  The change also makes
the fallback behaviour of sending the PROGRESS message better by sending
it only if dialtone is expected.

* Changed receiving a PROCEEDING message to not generate an
AST_CONTROL_PROGRESS frame if the progress indication ie indicates
non-end-to-end-ISDN.  This helps interoperability with SIP.

* Changed sending a PROCEEDING message in response to an
AST_CONTROL_PROCEEDING frame to not indicate inband audio available.  It
was silly to do so anyway because the channel driver doesn't know if
inband audio is even available.  This helps interoperability with SIP.

This patch and a corresponding change in libpri work together to allow
Asterisk to control the inband audio available progress indication ie on
the SETUP_ACKNOWLEDGE message when dialtone is present.

AST-1338 #close
Reported by: Tyler Stewart

Review: https://reviewboard.asterisk.org/r/3521/
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Merged revisions 413714 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413765 f38db490-d61c-443f-a65b-d21fe96a405b
rtp: Fix case typo in H263+ mime.

http://tools.ietf.org/html/rfc3555#section-4.2.6 says the canonical
mime subtype is "H263-1998", not "h263-1998". Original code was added
in r183101 on 2009-03-19 02:26:50 +0100.

This fixes issues with Polycom phones.

ASTERISK-23665 #close
ASTERISK-23665 #comment Patch r3529.patch uploaded by Guillaume Maudoux, backported by me.
Review: https://reviewboard.asterisk.org/r/3529/
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Merged revisions 413787 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413788 f38db490-d61c-443f-a65b-d21fe96a405b
h264: Fix H264 SDP payload format.

https://tools.ietf.org/html/rfc3984#section-8.1 says profile-level-id
takes 3 bytes in base16 (6 hex digits).

This fixes video setup in certain cases.

ASTERISK-23664 #close
ASTERISK-23664 #comment Patch r3530.patch uploaded by Guillaume Maudoux.
Review: https://reviewboard.asterisk.org/r/3530/


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413791 f38db490-d61c-443f-a65b-d21fe96a405b
chan_sip+CEL: Add missing ANSWER and PICKUP events to INVITE/w/replaces pickup.

When doing a "BLF-style call pickup" -- an INVITE with Replaces: header -- the
CEL log would lack the ANSWER and PICKUP events.

This patch adds the two missing events to the handle_invite_replaces() function.

ASTERISK-22977 #close
Review: https://reviewboard.asterisk.org/r/3073/
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Merged revisions 413832 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413838 f38db490-d61c-443f-a65b-d21fe96a405b
chan_sip: Add TLS and SRTP status to CLI command 'sip show channel'

ASTERISK-23564 #close
Reported by: Patrick Laimbock
Review: https://reviewboard.asterisk.org/r/3474/


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413876 f38db490-d61c-443f-a65b-d21fe96a405b
res_musiconhold: Minor cleanup.

Fix a few free()'s that should be ast_free()'s. Reverted an old
workaround that isn't necessary. Reorder a tiny bit of code.
Remove a bit of commented-out code.

Review: https://reviewboard.asterisk.org/r/3536/
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Merged revisions 413894 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: http://svn.asterisk.org/svn/asterisk/branches/11@413895 f38db490-d61c-443f-a65b-d21fe96a405b