While there were events defined for playback and recording
these were not actually sent. This change implements the
to_json handlers which produces them.

(closes issue ASTERISK-22710)
Reported by: Jonathan Rose

Review: https://reviewboard.asterisk.org/r/3026/
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Merged revisions 403119 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403120 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The Snoop operation can be invoked on a channel to spy or
whisper on it. It returns a channel that any channel operations
can then be invoked on (such as record to do monitoring).

(closes issue ASTERISK-22780)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3003/
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Merged revisions 403117 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403118 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In the last release of sounds, 1.4.25 we added a vm-msgforwarded prompt for various core languages. Now we use that prompt.

(issue ASTERISK-21413)
(closes issue ASTERISK-21413)
Reported by: netwrkr
Tested by: newtonr


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403106 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This fixes the unit tests that were broken by r403069 and several
functions requiring a new parameter for sanitization of JSON messages
generated from object snapshots.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403095 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Updated the alembic script for pjsip.  Also, the dtls config parsing stuff was
expecting strings with no underscores, so removed the underscores from the
option name before passing it to the parser.
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Merged revisions 403082 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403083 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change prevents channels used as implementation details from
leaking out to ARI. It does this by preventing creation of JSON blobs
of channel snapshots created from those channels and sanitizing JSON
blobs of bridge snapshots as they are created. This introduces a
framework for excluding information from output targeted at Stasis
applications on a consumer-by-consumer basis using channel sanitization
callbacks which could be extended to bridges or endpoints if necessary.

This prevents unhelpful error messages from being generated by
ast_json_pack.

This also corrects a bug where BridgeCreated events would not be
created.

(closes issue ASTERISK-22744)
Review: https://reviewboard.asterisk.org/r/2987/
Reported by: David M. Lee
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Merged revisions 403069 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403070 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Renamed, where appropriate, the configuration options for chan/res_pjsip to use
snake case (compound words separated by an underscore).  For example, faxdetect
will become fax_detect, recordofffeature will become record_off_feature, etc...

Review: https://reviewboard.asterisk.org/r/3002/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403051 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When translating from one format to another it is possible
to inform the translation function that the source frame should
be freed. This was previously done immediately but shortly
afterwards the frame that was freed was accessed and used again.

This change moves code around a bit so that the frame is now
freed after it has been completely used.

(closes issue ASTERISK-22788)
Reported by: Corey Farrell
Patches:
	translate-access-after-free-11up.patch uploaded by coreyfarrell (license 5909)
	translate-access-after-free-1.8.patch uploaded by coreyfarrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403017 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made PickupChan() search by channel uniqueids if the search could not
find a channel by name.

* Ensured PickupChan() never considers the picking channel for pickup.

* Made PickupChan() option p use a common search by name routine.  The
original search was erroneously case sensitive.

(issue AFS-42)

Review: https://reviewboard.asterisk.org/r/3017/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403013 65c4cc65-6c06-0410-ace0-fbb531ad65f3

By the time the directory application exits, a channel variable
DIRECTORY_RESULT will be set for the channel that invoked it which
can be used to determine the reason for exit. The changes log and
the app_directory documentation contain specific details about
each of the possible values for DIRECTORY_RESULT.

Review: https://reviewboard.asterisk.org/r/3016/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402995 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402994 65c4cc65-6c06-0410-ace0-fbb531ad65f3

For the new deviceState resource, we need to properly generate
device_state.[ch] files.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402982 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The direct media format capabilities are always allocated in
ast_sip_session_alloc and were not freed in the session destructor. Whoops.

(This being the third whoops caught by Scott and Nitesh's valgrind work for
the Asterisk Test Suite. Nifty!)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402969 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402957 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402945 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In PJMEDIA, pjmedia_sdp_rtpmap_to_attr will attempt to use the string
rtpmap.param regardless of its length value. Simply setting the length to 0
does not prevent the garbage on the stack in rtpmap.param.ptr from being
formatted in a sprintf call. This patch initializes the string to NULL so that
at the very least, something is provided to the function that is predictable.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402943 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes a reference counting memory leak on the ao2_container
created as part of create_mwi_subscriptions. When we create the container
in this routine, the intent is to hand lifetime ownership over to the global
container unsolicited_mwi. When ao2_global_obj_replace_unref is called, the
reference count on mwi_subscriptions (the container) will be bumped by 1;
however, the function does not decrement the reference count on
mwi_subscriptions when this occurs. This will prevent the container from being
fully disposed of when Asterisk exits (or on any subsequent call to this
operation, such as during a reload).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402942 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402929 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds the ability to start a silence generator on a channel
via ARI. This generator will play silence on the channel (avoiding audio
timeouts on the peer) until it is stopped, or some other media operation
is started (like playing media, starting music on hold, etc.).

(closes issue ASTERISK-22514)
Review: https://reviewboard.asterisk.org/r/3019/
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Merged revisions 402926 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402928 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The fromuser option is used to explicitly set the user within the From header. The
res_pjsip_caller_id module did not take this setting into account when determining
if the From header could be modified or not.

(closes issue ASTERISK-22866)
Reported by: Anthony Messina
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Merged revisions 402891 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402892 65c4cc65-6c06-0410-ace0-fbb531ad65f3

SIP transaction group lock support has been backported into our pjproject. Since the code
now internally uses a group lock the code is now changed to unlock it if present. Note
that the act of finding the transaction is what actually returns it locked.

For further information about group locks check out the wiki page at:
http://trac.pjsip.org/repos/wiki/Group_Lock

(issue ASTERISK-22818)
Reported by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402865 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/3008/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402854 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This fixes a crash when CELGenUserEvent is called from the dialplan
while CEL is disabled. Currently, CEL does not create its topics and
forwards if it is not enabled and external entities may depend on
these topics blindly since they should always be available. This patch
breaks up route creation and topic/forward creation such that the CEL
topics and forwards will always exist while the router and its
associated routes will be torn down and recreated as necessary.

(closes issue ASTERISK-22799)
Review: https://reviewboard.asterisk.org/r/3010/
Reported by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402839 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made Pickup() and PickupChan() tollerate empty pickup values.  i.e., You
can now have Pickup(&&exten@context).

* Made PickupChan() use the standard option flag parsing code.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402829 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402820 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Similar to how background works, if a say application is called with
this variable set to 'true', 'yes', 'on', etc. then using DTMF while
the say action is in progress will result in the channel jumping to
that extension in the dialplan.

Review: https://reviewboard.asterisk.org/r/3011/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402819 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Using the 'ring' operation it is possible to start locally generated ringback if
the channel is answered. This change adds the ability to stop it by using DELETE.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402805 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Was returning a 404 on a valid technology with an empty list of endpoints.
Now checking against the channel tech to make sure the tech itself is valid
and not just an empty list of endpoints.

(issue ASTERISK-22803)
Reported by: David M. Lee
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Merged revisions 402793 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402795 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Implementation listing endpoints by technology returned an empty array if no
matching endpoints were found.  Fixed so a "404 Not Found" will be returned
instead.

(closes issue ASTERISK-22803)
Reported by: David M. Lee
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Merged revisions 402787 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402788 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402778 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Two variables were being checked for NULLity immediately
after being declared NULL. I moved the NULL check until
after the variables are allocated.

This allows for the "channelvars" option in manager.conf
to work as intended again.
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Merged revisions 402767 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402768 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Both res_pjsip_messaging and res_pjsip_header_funcs were causing asterisk to
crash because they were trying to dereference a NULL pointer.

In the case of res_pjsip_messaging it was attempting to "print" a contact
header that did not exist.  In fact contact headers should not be part of
a SIP MESSAGE, so the offending code was simply removed.

In the case of res_pjsip_header_funcs a null private channel tech was being
passed to the function and then later dereferenced.  Added null checks (and
error logging) to the read/write function handlers to guard against crashing.

(closes issue ASTERISK-22821)
Reported by: Anthony Messina
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Merged revisions 402757 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402758 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This prevents NULL from being passed into an ast_json_pack call when no
extra information is passed to the application which prevents an error
message about NULL arguments from being generated.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402756 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402741 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Asterisk will sometimes core dump during caller id read on analog
channels due to a negative return value from the read() in
my_get_callerid that slips through as a negative length argument to
callerid_feed() if the errno returned by DAHDI is ELAST. This change
ensures that the negative return is treated properly even when it is
ELAST.

(closes issue ASTERISK-22746)
Reported by: Michael Walton
Patches:
    chan_dahdi_cid_crash_fix.r401410.patch uploaded by Michael Walton (License 6502)
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Merged revisions 402708 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 402709 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402711 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds a couple of test events for conference menu actions so that it's
easy to discern when those menu actions have been triggered.

(issue ASTERISK-22760)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2999/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402698 65c4cc65-6c06-0410-ace0-fbb531ad65f3

I'm doing some unrelated work in app_confbridge and finding
these "invalid pin" comments to be annoying. Get out!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402688 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In the current app_queue code from 1.8 up to trunk the upper and lower
penalties can be set to 0 but the value is interpreted to be disabled
instead of actually setting limits. This is especially evident if min
and max limits are set to 0 and members with penalties of 0 and 1 are
in the queue since the member with penalty 1 will still receive calls.
This patch adjusts the special disabled value to be INT_MAX instead of
0.

(closes issue ASTERISK-20862)
Review: https://reviewboard.asterisk.org/r/2995/
Reported by: Schmooze Com
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402648 65c4cc65-6c06-0410-ace0-fbb531ad65f3

For outbound register requests the tag on the From line was
updated every 20 seconds prior to a successful registration
and also once for each registration renewal.  That behavior
can possibly cause the registration to be denied because of
the different tag, and is not aligned with the intention of
RFC 3261 8.1.3.5 "... request constitutes a new transaction
and SHOULD have the same value of the Call-ID, To, and From
of the previous request...".  This updates chan_sip to have
a field to keep the local tag in the registration structure
and use that tag for registration requests where the callid
is also unchanged.

(closes issue ASTERISK-12117)
Reported by: Pawel Pierscionek
Review: https://reviewboard.asterisk.org/r/2988/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402607 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fix unlinking from the app_bridges_moh container in remove_bridge_moh()
without a lock under normal circumstances.

* Made check ast_bridge_set_after_callback() return value in
bridge_moh_create() to handle failure.

* Fixed SCOPED_AO2LOCK() locking over too much scope in
stasis_app_bridge_moh_channel() and stasis_app_bridge_moh_stop().

* Fixed unusual usage of ao2_unlink_flag() in control_unlink().

* Fixed orphaned bridge from off nominal path in
stasis_app_bridge_create().

* Fixed strange construct in stasis_app_unsubscribe().  From a bad merge?

* Made load_module() cleanup on failure.

Review: https://reviewboard.asterisk.org/r/2962/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402595 65c4cc65-6c06-0410-ace0-fbb531ad65f3