correct condition to check rtptimeout in [general] config section

ASTERISK-24393 #close
Reported by:  Dmitry Melekhov
Tested by:  Dmitry Melekhov
Patches:
  ASTERISK-24393.patch
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425591 65c4cc65-6c06-0410-ace0-fbb531ad65f3

With MALLOC_DEBUG the /main/config config_basic_ops test was causing a
SEGV while doing an ast_category_delete in an ast_category_browse loop.
Apparently this never worked but was also never tested.  I removed the
test, added 2 notes to config.h indicating that it's not supported and
added a few lines of code to ast_category_delete to prevent the SEGV
should someone attempt it in the future.

Tested-by: George Joseph

Review: https://reviewboard.asterisk.org/r/4078/
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Tasks that were marked for pending deletion in the scheduler would be moved to
the cache for later reuse, but after being recycled the deleted mark wouldn't
be removed resulting in fresh tasks being deleted without reason... and
immediately moved back into the cache where they could be reused again. This
could cause horrendous things to happen in just about anything that used a
scheduler.

ASTERISK-24321 #close
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4071/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425505 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Based on feedback from Richard, I created an accessor for
res_phoneprov/ast_phoneprov_std_variable_lookup and added
load priority to AST_MODULE_INFO.

Tested-by: George Joseph
Tested-by: Richard Mudgett

Review: https://reviewboard.asterisk.org/r/4076/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425482 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fax gateway session objects can be re-used, causing the
same gateway session to be added to faxregistry.container
more than once.  This change causes fax_session_new to
remove the reserved session from the container before
it's id is changed, ensuring it's possible for the
session to be freed.

ASTERISK-24392 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4049/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425460 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Masquerades into and out of channels that are involved in a dial operation
don't create the expected dial end event.  The missing dial end event goes
against the model for things like CDRs and generating Dial end manager
actions and such.

There are four cases:

1) A channel masquerades into the caller channel.  The case happens when
performing a blonde transfer using the channel driver's protocol.

2) A channel masquerades into a callee channel.  The case happens when
performing a directed call pickup.

3) The caller channel masquerades out of dial.  The case happens when
using the Bridge application on the caller channel.

4) A callee channel masquerades out of dial.  The case happens when using
the Bridge application on a peer channel.

As it turned out, all four cases need to be handled instead of just the
first one.

ASTERISK-24237
Reported by: Richard Mudgett

ASTERISK-24394 #close
Reported by: Richard Mudgett

Review: https://reviewboard.asterisk.org/r/4066/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425456 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Remove reference to module providing reserved session after
adding a reference to the final module.  This re-reference
is done to ensure that module references are correct even
if the final session selects a different module than the
reserved session.

ASTERISK-18923 #close
Reported by: Grigoriy Puzankin
Review: https://reviewboard.asterisk.org/r/4048/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425419 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch provides the capability to manipulate templates and categories
with non-unique names via AMI.

Summary of changes:

GetConfig and GetConfigJSON: Added "Filter" parameter:  A comma separated list
of name_regex=value_regex expressions which will cause only categories whose
variables match all expressions to be considered.  The special variable name
TEMPLATES can be used to control whether templates are included.  Passing
'include' as the value will include templates along with normal categories.
Passing 'restrict' as the value will restrict the operation to ONLY templates.
Not specifying a TEMPLATES expression results in the current default behavior
which is to not include templates.

UpdateConfig: NewCat now includes options for allowing duplicate category
names, indicating if the category should be created as a template, and
specifying templates the category should inherit from.  The rest of the
actions now accept a filter string as defined above.  If there are non-unique
category names, you can now update specific ones based on variable values.

To facilitate the new capabilities in manager, corresponding changes had to be
made to config, most notably the addition of filter criteria to many of the
APIs.  In some cases it was easy to change the references to use the new
prototype but others would have required touching too many files for this
patch so a wrapper with the original prototype was created.  Macros couldn't
be used in this case because it would break binary compatibility with modules
such as res_digium_phone that are linked to real symbols.

Tested-by: George Joseph

Review: https://reviewboard.asterisk.org/r/4033/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425363 65c4cc65-6c06-0410-ace0-fbb531ad65f3

After a reINVITE glare situation, Asterisk would re-send the reINVITE
even though the call had been hung up in the mean time.  This patch
unschedules the reinvite when handling the BYE.

ASTERISK-22791 #close
Reported by: Paolo Compagnini
Tested by: Paolo Compagnini

Review: https://reviewboard.asterisk.org/r/4056/
(testcase is in review r4055)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425300 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The main Makefile has a target test called 'badshell' that tests if
DESTDIR does not happen to have an an-expanded tilde (~).  This might
be the case if you run: make install DESTDIR=~/somewhere/

That test also disallowed valid tildes in directory names. The test is
now changed to only trigger on a tilde at the start of the path.

ASTERISK-13797 #close
Reported by: Tzafrir Cohen

Review: https://reviewboard.asterisk.org/r/4064/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425295 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Allow res_calendar_ews to work not only with libneon-0.29 but also
with 0.30.

ASTERISK-24325 #close
Reported by: Tzafrir Cohen

Review: https://reviewboard.asterisk.org/r/4068/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425290 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Tested module load/reload interaction between res_phoneprov and
res_pjsip_phoneprov_provider in cases where res_phoneprov didn't
load correctly (usually misconfiguration or missing phoneprov.conf)

Tested-by: George Joseph

Review: https://reviewboard.asterisk.org/r/4069/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425266 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When a smart bridge operation occurs and a bridge transitions from one
technology to another the old technology is provided the channels formerly
in it and told that they are leaving. Unfortunately the bridge provided
along with them is incomplete. The bridge, despite there being channels in it,
contains none. This forces technology implementations to have additional
logic when channels are leaving or to store their own duplicated
state.

This change makes the bridge more complete so it contains the expected
channels. Now that the bridge is complete special logic within
bridge_native_rtp is no longer needed and has been removed.

Review: https://reviewboard.asterisk.org/r/4057/
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If res_phoneprov failed to fully load (due to not being configured), the
providers container will be NULL. If a module attempts to register a phone
provisioning provider, it should check for the presence of the container.
If there is no providers container, it should return an error.

This patch makes the ast_phoneprov_provider_register function do that...
otherwise this would be a silly commit message.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425218 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This fixes a regression in callerid parsing introduced when another bug
was fixed. This bug occurred when the name was composed entirely of
DTMF keys and quoted without a number section (<>). 

ASTERISK-24406 #close
Reported by: Etienne Lessard
Tested by: Etienne Lessard
Patches:
    callerid_fix.diff uploaded by Kinsey Moore
Review: https://reviewboard.asterisk.org/r/4067/
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When the 'force_rport' option is enabled the behavior should be the same
as if the remote side placed rport into the message themselves. Therefore
any responses we send should include the source port of the request in the
rport of the Via header.

#SIPit31

ASTERISK-24387 #close
Reported by: Matt Jordan
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If a device re-INVITEs at the same time as the dialog is hung up, and
if then the ACK to the re-INVITE never reaches Asterisk, chan_sip would
fail to destroy the dialog after a while.  This resulted in (most
prominently) file handle leaks.

(Patch reindented by me.)

ASTERISK-20784 #close
ASTERISK-15879 #close
Reported by: Torrey Searle, Nitesh Bansal
Patches:
  reinvite_ack_timeout.patch uploaded by Torrey Searle (License #5334)
  patch_asterisk_20784.txt uploaded by Nitesh Bansal (License #6418)

Reviewboard: https://reviewboard.asterisk.org/r/4052/
(testcase can be found at r4051)
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endpoint->inbound_auths was changed to a vector in 13 and I
committed the 12 patch instead of the 13 patch.

Tested-by: George Joseph
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425053 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When starting ice if there is not at least one remote ice candidate with an RTP
component asterisk will crash. This is due to an assertion in pjnath as it
expects at least one candidate with an RTP component. Added a check to make
sure at least one candidate contains an RTP component and at least one candidate
has an RTCP component.

ASTERISK-24383 #close
Review: https://reviewboard.asterisk.org/r/4039/
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This module allows res_pjsip to integrate with res_phoneprov.  It handles
the pjsip 'phoneprov' object type.

Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/3976/
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This patch makes res_phoneprov more modular so other modules (like pjsip)
can provide configuration information instead of res_phoneprov relying solely
on users.conf and sip.conf.  To accomplish this a new ast_phoneprov public API
is now exposed which allows config providers to register themselves, set
defaults (server profile, etc) and add user extensions.

* ast_phoneprov_provider_register registers the provider and provides callbacks
  for loading default settings and loading users.
* ast_phoneprov_provider_unregister clears the defaults and users.
* ast_phoneprov_add_extension should be called once for each user/extension
  by the provider's load_users callback to add them.
* ast_phoneprov_delete_extension deletes one extension.
* ast_phoneprov_delete_extensions deletes all extensions for the provider.

Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/3970/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424965 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Now "cdr set debug on" doesn't also require "core set verbose 1" to see
CDR debug output.
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On systems with lots of RAM (e.g. 24GB) /proc/sys/fs/file-max divided
by two can exceed the per-process file limit of 2^20. This patch
ensures the value is capped.

(Patch cleaned up by me.)

ASTERISK-24011 #close
Reported by: Michael Myles
Patches:
  safe_asterisk-ulimit.diff uploaded by Michael Myles (License #6626)
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The underlying library, pjnath, that res_rtp_asterisk uses for ICE
support does not have support for ICE-TCP. As candidates are
passed through directly to it this can cause error messages to occur
when it receives something unexpected (such as a TCP candidate).
This change merely ignores all non-UDP candidates so they never
reach pjnath.

ASTERISK-24326 #close
Reported by: Joshua Colp
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This error message primarily applies to development tasks and will now
only show up when dev-mode is enabled via configure.
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In Asterisk 13+, any given message type is not guaranteed to exist even
if Asterisk comes up correctly since creation of the message type could
be declined. The indexer should not prevent Asterisk from starting
under these conditions.
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When message type creation is declined via stasis.conf, certain
operations log errors assuming that the declined type is being used
before initialization or after destruction. These error messages get
quite spammy for oft used message types and should not be logged in the
first place since the message type is validly NULL.

Reported by: Matt DiMeo
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Formats within a capabilities structure are addressed starting at 0, not 1.
Assuming 1 causes it to exceed an array.

ASTERISK-24389 #close
Reported by: Kevin Harwell
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Prior to this patch, the auth_reject_permanent parameter was not initialized on
the registration client state, leading to the parameter being disabled
regardless of the value specified in pjsip.conf.

This patch initialized the setting on the registration client state to the
provided configuration value.

ASTERISK-24398 #close
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If SendMessage encounters an error (such as incorrect input provided to the
action), it will currently return -1. Actions should only return -1 if the
connection to the AMI client should be closed. In this case, SendMessage
causing the client to disconnect is inappropriate.

This patch causes the action to return 0, which simply causes the action to
fail.

Review: https://reviewboard.asterisk.org/r/4024

ASTERISK-24354 #close
Reported by: Peter Katzmann
patches:
  sendMessage.patch uploaded by Peter Katzmann (License 5968)
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Using the Bridge application to bridge a channel that is executing an
applicaiton such as Wait results in a lingering Surrogate channel in the
CLI "core show channels" output even though it has already hungup.

* Fix bridge_exec() to not hold onto the current_dest_chan ref once it has
been put into the bridge.

* Eliminated bridge_exec()'s use of RAII_VAR().

ASTERISK-24224 #close
Reported by: Mark Michelson

Review: https://reviewboard.asterisk.org/r/4041/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424648 65c4cc65-6c06-0410-ace0-fbb531ad65f3

An OPTIONS request that is sent to Asterisk but not to a specific endpoint is
currently sent a 404 in response. This is because, not surprisingly, an empty
extension is never going to be found in the dialplan.

This patch makes it so that we only attempt to look up the endpoint in the
dialplan if it is specified in the OPTIONS request URI.

#SIPit31

ASTERISK-24370 #close
Reported by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424626 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your health.
It will treat the channels as a PJSIP channel, eventually hitting an ao2 error,
FRACKing on assertion error, and quite likely crashing.

This patch adds checks to the read/write callbacks that ensure that the channel
technology is of type 'PJSIP' before attempting to operate on the channel.

#SIPit31

ASTERISK-24382 #close
Reported by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424623 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When a message that exceeds the PJ_MAX_PKT_SIZE is sent over a reliable
transport, it is possible (although it shouldn't occur) for pjproject to pass
up an rdata object with a NULL msg in the msg_info. Needless to say, things
that attempt to dereference this are in for a rough ride.

In particular, this caused crashes in three different locations, all of which
are 'low level' enough to intercept an rdata object early in processing:

(1) res_pjsip_logger
(2) res_hep_pjsip
(3) res_pjsip/distributor

Anything that can intercept an rdata object before res_pjsip/distributor should
be defensive when looking at the received packet.

#SIPit31

ASTERISK-24369 #close
Reported by: Matt Jordan
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A subscription that has been persisted can - for various reasons - fail to be
re-created on startup. This patch resolves a number of crashes that occurred
when a subscription cannot be re-created on several off-nominal paths.

#SIPit31

ASTERISK-24368 #close
Reported by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424602 65c4cc65-6c06-0410-ace0-fbb531ad65f3