Some ast_stun_request users do not provide a destination address when
sending to a connection-mode socket.

ASTERISK-29691

Change-Id: Idd9114c3380216ba48abfc3c68619e79ad37defc

Without this message, it is not obvious that the reason is STUN timeout.

ASTERISK-29507 #close

Change-Id: I26e4853c23a1aed324552e1b9683ea3c05cb1f74

Added debug logging categories that allow a user to output debug
information based on a specified category. This lets the user limit,
and filter debug output to data relevant to a particular context,
or topic. For instance the following categories are now available for
debug logging purposes:

  dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet,
  stun, stun_packet

These debug categories can be enable/disable via an Asterisk CLI command.

While this overrides, and outputs debug data, core system debugging is
not affected by this patch. Statements still output at their appropriate
debug level. As well backwards compatibility has been maintained with
past debug groups that could be enabled using the CLI (e.g. rtpdebug,
stundebug, etc.).

ASTERISK-29054 #close

Change-Id: I6e6cb247bb1f01dbf34750b2cd98e5b5b41a1849

* netsock2.c: Test the addr->len member first as it may be the only member
initialized in the struct.

* stun.c:ast_stun_handle_packet(): The combinded[] local array could get
used uninitialized by ast_stun_request().  The uninitialized string gets
copied to another location and could overflow the destination memory
buffer.

These valgrind findings were found for ASTERISK_27150 but are not
necessarily a fix for the issue.

Change-Id: I55f8687ba4ffc0f69578fd850af006a56cbc9a57

If ast_stun_request() receives packets other than a STUN response then we
could conceivably never exit if we continue to receive packets with less
than three seconds between them.

* Fix poll timeout to keep track of the time when we sent the STUN
request.  We will now send a STUN request every three seconds regardless
of how many other packets we receive while waiting for a response until we
have completed three STUN request transmission cycles.

Change-Id: Ib606cb08585e06eb50877f67b8d3bd385a85c266

ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.

Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename

This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled.  This variable was only used in lock.c so it
is now initialized in that file only.

ASTERISK-26480 #close

Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966

Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.

Specifically, it does the following:

* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
  remove passing the version in with the macro. Other facilities
  than 'core show file version' make use of the file names, such as
  setting a debug level only on a specific file. As such, the act of
  registering source files with the Asterisk core still has use. The
  macro rename now reflects the new macro purpose.

* main/asterisk:
  - Refactor the file_version structure to reflect that it no longer
    tracks a version field.
  - Remove the "core show file version" CLI command. Without the file
    version, it is no longer useful.
  - Remove the ast_file_version_find function. The file version is no
    longer tracked.
  - Rename ast_register_file_version/ast_unregister_file_version to
    ast_register_file/ast_unregister_file, respectively.

* main/manager: Remove value from the Version key of the ModuleCheck
  Action. The actual key itself has not been removed, as doing so would
  absolutely constitute a backwards incompatible change. However, since
  the file version is no longer tracked, there is no need to attempt to
  include it in the Version key.

* UPGRADE: Add notes for:
  - Modification to the ModuleCheck AMI Action
  - Removal of the "core show file version" CLI command

Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e

Since 'core stop now' and 'core restart now' do not stop modules,
it is unsafe for most of the core to run cleanups.  Originally all
cleanups used ast_register_atexit, and were only changed when it
was shown to be unsafe.  ast_register_atexit is now used only when
absolutely required to prevent corruption and close child processes.

Exceptions that need to use ast_register_atexit:
* CDR: Flush records.
* res_musiconhold: Kill external applications.
* AstDB: Close the DB.
* canary_exit: Kill canary process.

ASTERISK-24142 #close
Reported by: David Brillert

ASTERISK-24683 #close
Reported by: Peter Katzmann

ASTERISK-24805 #close
Reported by: Badalian Vyacheslav

ASTERISK-24881 #close
Reported by: Corey Farrell

Review: https://reviewboard.asterisk.org/r/4500/
Review: https://reviewboard.asterisk.org/r/4501/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433498 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When sending the USERNAME attribute in an RTP STUN
response, the implementation in append_attr_string
passed the actual length, instead of padding it up
to a multiple of four bytes as required by the RFC
3489.  This change adds separate variables for the
string and padded attributed lengths, and performs
padding correctly.

Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/4139/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427877 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377903 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
      unregister-cli-multiple-all.patch (license #5909) patch uploaded by Corey Farrell
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377884 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When receiving a STUN binding request send one out as the Google Talk client uses this as a method to determine if the remote party is still reachable or not.

Failure to do this results in the Google Talk client ignoring RTP packets after a specific period of time. This is also done as a result of receiving a STUN binding request so that the username information can be used from the inbound request, thus not requiring it to be stored on a per candidate basis.

(closes issue ASTERISK-20107)
Reported by: Malcolm Davenport


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369858 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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  r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines
  
  Add support-level indications to many more source files.
  
  Since we now have tools that scan through the source tree looking for files
  with specific support levels, we need to ensure that every file that is
  a component of a 'core' or 'extended' module (or the main Asterisk binary)
  is explicitly marked with its support level. This patch adds support-level
  indications to many more source files in tree, but avoids adding them to
  third-party libraries that are included in the tree and to source files
  that don't end up involved in Asterisk itself.
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  r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines
  
  Add a script to enable finding source files without support-levels defined.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369013 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The STUN socket must remain open between polls or the external address
seen by the STUN server is likely to change.  However, if the STUN request
poll fails then the STUN server address needs to be re-resolved and the
STUN socket needs to be closed and reopened.

* Re-resolve the STUN server address and create a new socket if the STUN
request poll fails.

* Fix ast_stun_request() return value consistency.

* Fix ast_stun_request() to check the received packet for expected message
type and transaction ID.

* Fix ast_stun_request() to read packets until timeout or an associated
response packet is found.  The stun_purge_socket() hack is no longer
required.

* Reduce ast_stun_request() error messages to debug output.

* No longer pass in the destination address to ast_stun_request() if the
socket is already bound or connected to the destination.

(closes issue ASTERISK-18327)
Reported by: Wolfram Joost
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/1595/
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https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r284597 | tilghman | 2010-09-02 00:00:34 -0500 (Thu, 02 Sep 2010) | 29 lines
  
  Merged revisions 284593,284595 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
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    r284593 | tilghman | 2010-09-01 17:59:50 -0500 (Wed, 01 Sep 2010) | 18 lines
    
    Merged revisions 284478 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
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      r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) | 11 lines
      
      Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows.
      
      This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing
      a potential crash bug in all supported releases.
      
      (closes issue #17678)
       Reported by: russell
      Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select 
      
      Review: https://reviewboard.asterisk.org/r/824/
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    r284595 | tilghman | 2010-09-01 22:57:43 -0500 (Wed, 01 Sep 2010) | 2 lines
    
    Failed to rerun bootstrap.sh after last commit
  ................
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284598 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The channel drivers which have been most heavily tested with these enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
introduced in a later commit. chan_skinny has code added to it here, but according
to user pj, the support on chan_skinny is not working as of now. This will be fixed in
a later commit.

A special thanks goes out to bugtracker user gareth for getting the ball rolling and
providing the initial support for this work. Without his initial work on this, this would
not have been nearly as painless as it was.

This functionality has been tested by Digium's product quality department, as well as a
customer site running thousands of calls every day. In addition, many many many many bugtracker
users have tested this, too.

(closes issue #8824)
Reported by: gareth

Review: http://reviewboard.digium.com/r/201



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.

Review: http://reviewboard.digium.com/r/209/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3