This patch allows individual dialplan functions to be marked as
'dangerous', to inhibit their execution from external sources.

A 'dangerous' function is one which results in a privilege escalation.
For example, if one were to read the channel variable SHELL(rm -rf /)
Bad Things(TM) could happen; even if the external source has only read
permissions.

Execution from external sources may be enabled by setting
'live_dangerously' to 'yes' in the [options] section of asterisk.conf.
Although doing so is not recommended.

Also, the ABI was changed to something more reasonable, since Asterisk
12 does not yet have a public release.

(closes issue ASTERISK-22905)
Review: http://reviewboard.digium.internal/r/432/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403960 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The ast_bridge_set_transfer_variables function is supposed to wipe whichever
variable isn't being set. Instead it was setting both to the new value.  Oops.

(issue AFS-24)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403958 65c4cc65-6c06-0410-ace0-fbb531ad65f3

During dialplan execution in pbx_extension_helper(), the contexts global
read lock prevents link list corruption, but was released with a pointer
to the ast_exten and data later used in variable substitution.  Instead,
this patch removes pbx_substitute_variables() and locates a copy of the
ast_exten data on the stack before releasing the lock, where ast_exten
could get free'd by another thread performing a module reload.

(issue AST-1179)
Reported by: Thomas Arimont
(issue AST-1246)
Reported by: Alexander Hömig
Review: https://reviewboard.asterisk.org/r/3055/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403865 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch prevents an infinite loop overwriting memory when
a message is received into the unpacksms16() function, where
the length of the message is an odd number of bytes.

(closes issue ASTERISK-22590)
Reported by: Jan Juergens
Tested by: Jan Juergens
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403857 65c4cc65-6c06-0410-ace0-fbb531ad65f3

While entertaining, sizeof(buflen) is not the same as buflen. Doh.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403824 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Objects which are involved in SIP request creation and sending
now allow an outbound proxy to be specified. For cases where
an endpoint is used the outbound proxy specified there will
be applied.

(closes issue ASTERISK-22673)
Reported by: Antti Yrjola

Review: https://reviewboard.asterisk.org/r/3022/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403812 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change adds an event for when an originated call is redirected to
another target. This event contains the original channel and the newly
created channel. If a stasis subscription exists on the original originated
channel for a stasis application then a new subscription will also be
created on the stasis application to the redirected channel. This allows
the application to follow the call path completely.

(closes issue ASTERISK-22719)
Reported by: Joshua Colp

Review: https://reviewboard.asterisk.org/r/3054/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403810 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403797 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Rerunning a failed unit test after loading any required modules should
allow the test to report a pass status if it now passes.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403784 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There were still a few cases in which ATTENDEDTRANSFER and BLINDTRANSFER
wouldn't be set on channels involved with blind and attended transfers.
This would happen with features that were initialized by channel driver
specific mechanisms in multiparty calls. This patch resolves those cases
while attempted to keep the behavior for setting those variables as
consistent as possible.

(closes issue AFS-24)
Review: https://reviewboard.asterisk.org/r/3040/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403783 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The change contains a slightly adjusted patch that was on the issue
(submitted by kmoore).  A fix was made by adding in a bridge lock
while calling bridge_start/stop from the framehook callback.  Since
the framehook callback is not called from the bridging core the bridge
is not locked, but needs to be before calling bridge_start.

(closes issue ASTERISK-22749)
Reported by: Kinsey Moore
Review: https://reviewboard.asterisk.org/r/3066/
Patches:
     lock_inversion.diff uploaded by kmoore (license 6273)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403768 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added the ability to specify channel variables when creating/originating a
channel in ARI.  The variables are sent in the body of the request and should
be formatted as a single level JSON object.  No nested objects allowed.
For example: {"variable1": "foo", "variable2": "bar"}.

(closes issue ASTERISK-22872)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3052/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403757 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added the ability to have rules that are checked when adding and/or removing
channels to/from a bridge.  In this case, if a channel is currently recording
and someone attempts to add it to a bridge an "is recording" rule is checked,
fails, and a 409 conflict is returned.

Also command functions now return an integer value that can be descriptive of
what kind of problems, if any, occurred before or during execution.

(closes issue ASTERISK-22624)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2947/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403737 65c4cc65-6c06-0410-ace0-fbb531ad65f3

It is much nicer diagnosing a test failure if app_voicemail is actually
loaded.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403726 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When the alembic scripts were written for creating Asterisk
realtime databases the extensions table for dialplan wasn't
included.  This update creates the extensions table.

(closes issue ASTERISK-22815)
Reported by: Zone Conkle
Review: https://reviewboard.asterisk.org/r/3064/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403714 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch was intended to eliminate a deadlock that occurs when
masquerades occur in pjsip channels, but has some potential side
effects. Mark Michelson is currently working on addressing this
problem from another angle.

(issue ASTERISK-22936)
Reported by: Jonathan Rose
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403707 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In some cases messages need to be sent to a direct URI (sip:<ip address>). This
patch adds in that support by using a default outbound endpoint.  When sending
messages, if no endpoint can be found then the default one is used.

To facilitate this a new default_outbound_endpoint option was added to the
globals section for pjsip.conf.

Review: https://reviewboard.asterisk.org/r/2944/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403687 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If you set a peer's outboundproxy and then removed it from the config,
this would not get picked up in a config reload.  This patch fixes that
by resetting it in set_peer_defaults().

Closes ASTERISK-19454
Review: https://reviewboard.asterisk.org/r/3065/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403652 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* The voicemail registration/unregistration functions now take a struct of
callbacks instead of a lengthy parameter list of callbacks.

* The voicemail registration/unregistration functions now prevent a
competing module from interfering with an already registered callback
supplying module.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403643 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds CHANNEL read support for chan_pjsip. This allows the dialplan
to use the CHANNEL function on a chan_pjsip channel to obtain run-time
information about the channel from the PJSIP channel driver and the PJSIP
stack. This includes:
 * RTP information, including source/destination media addresses, whether or
   not the media is secure, held, and other properties.
 * RTCP information. This includes sets of parseable information, as well as
   individual statistic attriutes.
 * PJSIP information. This includes URIs, local/remote signalling addresses,
   whether or not the signalling is secure, and other properties.
 * The endpoint name. This can be used in conjunction with the PJSIP_ENDPOINT
   function to obtain more detailed endpoint information.

Review: https://reviewboard.asterisk.org/r/3038/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403619 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds a new function, PJSIP_ENDPOINT, which lets the dialplan query,
for any endpoint, any property configured on an endpoint. This function is a
companion to the CHANNEL function, which can be used to extract the endpoint
name for a channel.

Review: https://reviewboard.asterisk.org/r/3035
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403617 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When switching to using a vector for authentication, I initialized
the vector for the artificial endpoint to be of size 1. However, this
does not result in AST_VECTOR_SIZE() returning 1 since there isn't
actually anything in the vector.

Rather than trifle with the vector by putting unnecessary elements in,
I simply changed the callback in res_pjsip_authenticator_digest.c to
explicitly report that the artificial endpoint requires authentication.

Thanks to Joshua Colp for pointing this out.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403605 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-22936)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/3042/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403588 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue AFS-14)
Review: https://reviewboard.asterisk.org/r/3045/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403576 65c4cc65-6c06-0410-ace0-fbb531ad65f3

res_sorcery_astdb.c: Fix get multiple records by regex.

* Fix sorcery_astdb_retrieve_regex() pattern matching.  Let the regexec()
function match the stored key values instead of having astdb prefilter
them.  Previoiusly you could only use a simple regex pattern when the
pattern began with '^'.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403560 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fix sorcery_astdb_retrieve_regex() pattern matching.  Let the regexec()
function match the stored key values instead of having astdb prefilter
them.  Previoiusly you could only use a simple regex pattern when the
pattern began with '^'.

* Fix off nominal memory leak in sorcery_astdb_retrieve_regex().
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403546 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Eliminated shadowing of the __ast_sorcery_apply_config() name parameter
causing confusion.

* Fix potential crash from sorcery.conf user input in
__ast_sorcery_apply_config() if the user supplied a malformed config line
that is missing the sorcery object type name.

* Remove redundant test in __ast_sorcery_apply_config().  !config and
config == CONFIGS_STATUS_FILEMISSING are identical.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403544 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The snapshot process for endpoints uses the channel ids present
on the endpoint itself. Without keeping a reference it was possible
for the strings to be freed underneath any consumer of an endpoint
snapshot.

A reference is now held by the snapshot to the channel ids and
released when the snapshot is destroyed.

(issue ASTERISK-22801)
Reported by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403543 65c4cc65-6c06-0410-ace0-fbb531ad65f3

You would think that a new file would start off without any whitespace
oddities.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403528 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/3009



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403526 65c4cc65-6c06-0410-ace0-fbb531ad65f3

For the time, this is only useful for retrieving the filename.

The purpose of this function is to better facilitate multiple
mixmonitors per channel. Setting a MIXMONITOR_FILENAME channel
variable is not conducive to such behavior, so allowing finer
grained access to individual mixmonitor properties improves
the situation. The MIXMONITOR_FILENAME channel variable is still
set, though, so there is no worry about backwards compatibility.

Review: https://reviewboard.asterisk.org/r/3023



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403512 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Due to the way pjproject internally works it was possible for the
NAT module to not be invoked on messages with-in a session dialog.
This means that the various parts of the message would not get rewritten
with the source IP address and port.

This change uses a session supplement to add the NAT module
to the dialog on the first incoming or outgoing INVITE.

(closes issue ASTERISK-22941)
Reported by: Leif Madsen
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403511 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Since Asterisk has a vector API now, places where arrays are manually
resized don't really make sense any more. Since the auth work in PJSIP
was freshly-written, it was easy to reform it to use a vector.

Review: https://reviewboard.asterisk.org/r/3044



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403499 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prior to this patch, res_fax_spandsp was conservative with how it initialized
the spandsp T.38 context. It would only initialize it if the driver thought
the current state was a T.38 fax. While this works fine in nominal situations,
in certain off nominal situations, res_fax_spandsp can believe that a T.38
fax will not occur when in fact one has started. In particular, this was
discovered when res_fax would fall back to audio after timing out on a T.38
upgrade. The SIP channel driver would continue to retry the re-INVITE and -
if the remote end responded after res_fax timed out with a 200 OK - a T.38
frame would be delivered to the res_fax stack when it no longer expected it.

As it turns out, there does not appear to be any downside to always
initializing the T.38 context, other than the actual memory allocation.
Since that avoids this off nominal situation (and others which are equally
likely hard to predict), this is the safest way to avoid this problem.

Much thanks to Torrey as well for providing a scenario that reproduces this
issue.

(closes issue ASTERISK-21242)
Reported by: Ashley Winters
Tested by: Torrey Searle
patches:
  always-init-t38.patch uploaded by awinters (License 6477)
  A_PARTY.xml uploaded by tsearle (License 5334)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403466 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If the CDR unregistration fails due to an inflight CDR, the
res_config_sqlite module needs to bail on unloading itself. Otherwise,
the config could be unloaded (including the CDR table name) while the
CDR engine posts a CDR to the still registered backend, resulting in
a crash.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403436 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Using this terminator when active results in ${RECORD_STATUS} being set to
'OPERATOR' instead of 'DTMF'

(closes issue AFS-7)

Review: https://reviewboard.asterisk.org/r/3041/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403414 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403404 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The code for getting channel variables from ARI assumed that you needed
to lock the channel in order to properly execute functions and read
channel variables. Apparently, this is not the case, since any dialplan
function that puts the channel into autoservice deadlocks when
attempting to remove the channel from autoservice.
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........
  r403304 | dlee | 2013-12-02 12:34:50 -0600 (Mon, 02 Dec 2013) | 1 line
  
  Fixed the filename for the ari.conf docs
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  r403310 | file | 2013-12-03 10:32:12 -0600 (Tue, 03 Dec 2013) | 5 lines
  
  Revert revision 403304: Fixed the filename for the ari.conf docs
  
  The changed value refers to the name of the module. The name of the
  configuration file is specified in the configFile section.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403402 65c4cc65-6c06-0410-ace0-fbb531ad65f3