Removed the exportation of global symbols from the module as it is no longer
needed and it could potentially cause load problems as on some systems it
would try to load before res_pjsip_pubsub
........

Merged revisions 407034 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407035 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made ChanSpy accept a channel uniqueid or a fully specified channel name
as the chanprefix parameter if the 'u' option is specified.

(closes issue AFS-42)

Review: https://reviewboard.asterisk.org/r/3160/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407033 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 407031 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407032 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When the PJSIP pubsub framework was created, subscription handlers were required
to state what event they handled along with what body types they knew how to
generate. While this serves well when implementing a base RFC, it has problems
when trying to extend the body to support non-standard or proprietary body
elements. The code also was NOTIFY-specific, meaning that when the time comes
that we start writing code to send out PUBLISH requests with MWI or presence
bodies, we would likely find ourselves duplicating code that had previously been
written.

This changeset introduces the concept of body generators and body supplements. A
body generator is responsible for allocating a native structure for a given body
type, providing the primary body content, converting the native structure to a
string, and deallocating resources. A body supplement takes the primary body
content (the native structure, not a string) generated by the body generator and
adds nonstandard elements to the body. With these elements living in their own
module, it becomes easy to extend our support for body types and to re-use
resources when sending a PUBLISH request.

Body generators and body supplements register themselves with the pubsub core,
similar to how subscription and publish handlers had done. Now, subscription
handlers do not need to know what type of body content they generate, but they
still need to inform the pubsub core about what the default body type for a
given event package is. The pubsub core keeps track of what body generators and
body supplements have been registered. When a SUBSCRIBE arrives, the pubsub core
will check that there is a subscription handler for the event in the SUBSCRIBE,
then it will check that there is a body generator that can provide the content
specified in the Accept header(s).

Because of the nature of body generators and supplements, it means
res_pjsip_exten_state and res_pjsip_mwi have been completely gutted. They no
longer worry about body types, instead calling
ast_sip_pubsub_generate_body_content() when they need to generate a NOTIFY body.

Review: https://reviewboard.asterisk.org/r/3150
........

Merged revisions 407016 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407030 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A couple of the scripts had errors that would not allow a full migration to
take place.  The extensions table needed to make its 'id' column a primary
key in order to work with mysql.  The other script ...add_endpoints... was
missing tables that it was trying to add columns to.

Added the primary key on id for extensions and added the tables in for the
missing pjsip configuration options.  While it is not ideal to modify already
released scripts this was a case where it had to be done due to errors in
the script and lacking a better alternative.

Review: https://reviewboard.asterisk.org/r/3167/
........

Merged revisions 407019 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407029 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When subscribing to MWI (res_pjsip_mwi) and the sip uri did not contain a name
(ex: sip:<ip address>) then the subscription would fail since it would be unable
to locate an associated aor.  This patch makes it so that when a subscribe comes
with no aor name then it will subscribe to all aors on the located endpoint.

(closes issue ASTERISK-23072)
Reported by: Bob M
Review: https://reviewboard.asterisk.org/r/3164/
........

Merged revisions 407014 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407015 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In NAT scenarios where a call is placed to a Grandstream phone,
res_pjsip will sometimes send the ACK to a 200 OK to the private
address of the device behind the NAT instead of the address of the NAT
device. This corrects that behavior by rewriting the address in the
Contact header in the incoming 200 OK and the dialog's target address
if necessary (since it has already been rewritten to the incorrect
private address).

(closes issue ASTERISK-23106)
Review: https://reviewboard.asterisk.org/r/3168/
Reported by: Matt Jordan
........

Merged revisions 407000 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407001 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Return before chan is possibly unlocked a second time when hanging up
a channel in SUBSTATE_OFFHOOK.
........

Merged revisions 406987 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406988 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ast_bind to a port reserved for another program by SELinux causes
errno == EACCES.  This caused random failures when binding rtp or
udptl sockets.  Treat EACCES as a non-fatal error, try next port.

(closes issue ASTERISK-23134)
Reported by: Corey Farrell
........

Merged revisions 406933 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 406934 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 406935 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406936 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 406918 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 406919 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406920 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Closes issue ASTERISK-22662
........

Merged revisions 406860 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 406861 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 406862 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406863 65c4cc65-6c06-0410-ace0-fbb531ad65f3

What seems to be happening is if a subscription has been terminated and the
subscription timeout/expires is less than the time it takes for all pending
transactions (currently on the subscription) to end then the subscription
timer will not have been canceled yet and sub will be null.  Since the
subscription has already been canceled nothing needs to be done so a null
check in the asterisk code is sufficient in working around this problem.

(closes issue ASTERISK-23129)
Reported by: Dan Jenkins
........

Merged revisions 406847 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406848 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Asterisk's RADIUS module currently build against libradiusclient-ng, but this
project has been superseeded by libfreeradius-client. The API is 99% compatible
except that the header name has changed, the library name has changed, and
the configuration file location has changed.

(closes issue ASTERISK-22980)
Reported by: Jeremy Lainé
Patches:
     freeradius-client.patch uploaded by sharky (license 6561)
........

Merged revisions 406801 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 406802 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 406803 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406825 65c4cc65-6c06-0410-ace0-fbb531ad65f3

On some systems the values for INFINITY and NAN are not defined thus causing
a build error on those systems.  Added definitions for those if they had
not previously been defined.

(closes issue ASTERISK-23056)
Reported by: capouch
Patches:
     inf-nan-patch.txt uploaded by capouch (license 6564)
........

Merged revisions 406788 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406789 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Currently, attempting to subscribe an application to a device state
that it has already subscribed to will generate a 500 error response.
This will now be treated as a subscription refresh even though ARI
subscriptions don't currently support lifetimes and will respond with
the normal response for a successful subscription (200 OK).

(closes issue ASTERISK-23143)
Reported by: Matt Jordan
........

Merged revisions 406775 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406778 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In ast_rtp_instance_make_compatible(), after a failure of
channel tech call get_rtp_info() to return peer_instance,
the null pointer would be passed to ao2_ref, producing an
error that looked like a refernce counting problem but is
not.  This patch corrects that and adds helpful LOG_ERROR
messages to indicate which failure path occurred.

(issue AST-1276)
Review: https://reviewboard.asterisk.org/r/3156/
........

Merged revisions 406721 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 406722 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 406723 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406724 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fixed the test_cel_attended_transfer_bridges_link unit test to also
account for the local channel link being destroyed now that the bridges
are actually destroyed.

* Made CDR unit test use its own version of do_sleep() from the CEL unit
tests.
........

Merged revisions 406707 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406710 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added a note in the changes file about the new 'StatusText' field that was
added to the 'ExtensionStatus' event.

(issue ASTERISK-23154)
Reported by: Jonathan Rose


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406696 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When an 'ExtensionStatus' event was raised it included the status as a
numerical value, but did not include a text description of the status.
Added a 'StatusText' field to the event which is a string representation
of the extension status.  Also added this to the 'Extension State' command
response.

(closes issue ASTERISK-23154)
Reported by: Jonathan Rose


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406647 65c4cc65-6c06-0410-ace0-fbb531ad65f3

extconfig.conf was hard-coded to not allow nested includes for some reason.
The code has been this way since a patch was merged for ASTERISK-3333 (revision
4889), which was a significant update to this code ("Merge config updates").

I can't figure out any good reason why this should be limited.  This patch just
removes the limit and uses the default nesting depth limit.

Closes issue ASTERISK-17837

Review: https://reviewboard.asterisk.org/r/3159/
........

Merged revisions 406643 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 406644 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 406645 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406646 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In pre-trunk versions (...12) it accepts a basic regex, which is
confusing because all other regexes in asterisk are of the
extended kind.

Review: https://reviewboard.asterisk.org/r/3147/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406618 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The ast_filestream object gets tacked on to a channel via
chan->timingdata.  It's a reference counted object, but the reference
count isn't used when putting it on a channel.  It's theoretically
possible for another thread to interfere with the channel while it's
unlocked and cause the filestream to get destroyed.

Use the astobj2 reference count to make sure that as long as this code
path is holding on the ast_filestream and passing it into the file.c
playback code, that it knows it's valid.

Bug reported by Leif Madsen.

Review: https://reviewboard.asterisk.org/r/3135/
........

Merged revisions 406566 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 406567 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 406574 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406595 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 406514 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 406515 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 406516 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406517 65c4cc65-6c06-0410-ace0-fbb531ad65f3

live_ast can be used as a wrapper script to run asterisk, gdb or
valgrind. In those cases it runs them and returns the result. It is more
useful to use 'exec' to avoid having another odd process in the chain.

Review: https://reviewboard.asterisk.org/r/3110/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406503 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The core may (depending on circumstances) request a single codec on outgoing
calls. Many channel drivers ignore or treat this as a suggestion while still
including configured codecs. The res_pjsip_session logic treated this as
an explicit request, leaving out other configured codecs.

This change makes res_pjsip_session behave like other channel driver and simply
adds the requested codec to the list.

(closes issue ASTERISK-23082)
Reported by: xrobau

Review: https://reviewboard.asterisk.org/r/3140/
........

Merged revisions 406489 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406490 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The CEL data structures need to be protected during a configuration reload
and shutdown.  Asterisk crashed during a shutdown because CEL events were
still in flight and the CEL data structures were already destroyed.

* Protected the cel_backends, cel_dialstatus_store, and cel_linkedids ao2
containers with a global ao2 object wrapper.

* Added NULL checks before use of the cel_backends, cel_dialstatus_store,
and cel_linkedids ao2 containers in case the CEL module is already
shutdown.

* Fixed overloading of the cel_linkedids held objects reference count.
During shutdown any held objects would be leaked.

* Fixed memory leak of cel_linkedids held objects if the LINKEDID_END is
not being tracked.  The objects in the cel_linkedids container were not
removed if the LINKEDID_END event is not used.

* Added access protection to the cel_backends container during the CLI
"cel show status" command.

* Made cel_backends, cel_dialstatus_store, and cel_linkedids use the
standard ao2 callback templates for the hash and cmp functions.

* Eliminated unnecessary uses of RAII_VAR().

* Made ast_cel_engine_init() cleanup alocated resources on failure.

(closes issue AST-1253)
Reported by: Guenther Kelleter

Review: https://reviewboard.asterisk.org/r/3128/
........

Merged revisions 406417 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 406418 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 406465 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406466 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds the LWP to core show locks output if it is available.

Review: https://reviewboard.asterisk.org/r/3142/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406416 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made register atexit shutdown routine only once in __init_manager().

* Fixed some initial load failure conditions in __init_manager().

* Made reset options to defaults on reload when the reload will actually
happen.

* Removed unnecessary container traversals of the white/black filters
during manager_free_user().

* ast_free() does not need a NULL check before calling.
........

Merged revisions 406359 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 406400 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 406401 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406407 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/3141/
........

Merged revisions 406360 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 406361 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 406389 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406399 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Occasionally, the manager module would get an "INTERNAL_OBJ: bad magic
number" error on a "core restart gracefully" command if an AMI connection
is established.

* Added ao2_global_obj protection to the sessions global container.

* Fixed the order of unreferencing a session object in session_destroy().

* Removed unnecessary container traversals of the white/black filters
during session_destructor().

(closes issue AST-1242)
Reported by: Guenther Kelleter

Review: https://reviewboard.asterisk.org/r/3144/
........

Merged revisions 406341 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 406342 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406343 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 406327 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406328 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When support for a realtime sorcery module was added in revision 386731, the
wrong property was accidentally used for setting the column name to be updated
in the database table.  This patch fixes the typo.

(closes issue ASTERISK-23177)
Reported by: Denis
Tested by: Denis
Patches:
    asterisk-23177-use-field-name.diff by Michael L. Young (license 5026)
........

Merged revisions 406311 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406312 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
  r406294 | mmichelson | 2014-01-23 15:00:24 -0600 (Thu, 23 Jan 2014) | 11 lines
  
  Fix presence body errors found during testing:
  
  * PIDF bodies were reporting an "open" state in many cases where
    it should have been reporting "closed"
  * XPIDF bodies had XML nodes placed incorrectly within the hierarchy.
  * SIP URIs in XPIDF bodies did not go through XML sanitization
  * XML sanitization had some errors:
      * Right angle bracket was being replaced with "&rt;" instead of ">"
  	* Double quote, apostrophe, and ampersand were not being escaped.
........
  r406295 | mmichelson | 2014-01-23 15:09:35 -0600 (Thu, 23 Jan 2014) | 11 lines
  
  Fix presence body errors found during testing:
  
  * PIDF bodies were reporting an "open" state in many cases where
    it should have been reporting "closed"
  * XPIDF bodies had XML nodes placed incorrectly within the hierarchy.
  * SIP URIs in XPIDF bodies did not go through XML sanitization
  * XML sanitization had some errors:
      * Right angle bracket was being replaced with "&rt;" instead of ">"
  	* Double quote, apostrophe, and ampersand were not being escaped.
........

Merged revisions 406294-406295 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406298 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In ast_build_timing, initialize the timezone value to NULL
in order to avoid deferencing an uninitialized value later
when calling ast_destroy_timing.  The timezone value could
be uninitialized if ast_build_timing were to fail due to a
zero length time string.

(closes issue ASTERISK-22861)
Reported by: Sebastian Murray-Roberts
Review: https://reviewboard.asterisk.org/r/3134/
Patches:
     ast_build_timing-initialize-timezone.patch uploaded by coreyfarrell (license 5909)
........

Merged revisions 406241 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 406245 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 406264 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406269 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Confbridge AMI and CLI commands for mute, unmute, and setting the
single video source can accept channel prefixes in lieu of a full
channel name, but documentation states only that it is required and is
a channel name. This corrects the documentation.

(closes issue PQ-1397)
Reported by: Steve Pitts
........

Merged revisions 406217 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 406223 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406224 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change allows chan_sip to decline individual image streams over
unsupported transports in the SDP of the 200 response. Previously,
an image stream offer with RTP/AVP as the transport would cause
chan_sip to respond with a 488.

(closes issue ASTERISK-22988)
Reported by: adomjan
Original patch by: adomjan
........

Merged revisions 406170 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 406171 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 406172 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406173 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Several of the playback error messages for invalid media input in
res_stasis_playback.c had the media name and channel name reversed.
They now correctly identify the channel name and media name.

Reported by: skrusty
........

Merged revisions 406152 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406153 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Making the help text for both more explicit regarding the format of mailbox identifiers. i.e. clarifying the format for app_voicemail mailboxes vs mailboxes from external MWI sources through modules such as res_external_mwi.
........

Merged revisions 406133 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406134 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/3133/
........

Merged revisions 406079 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 406080 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 406081 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406082 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This restricts direct usage of global oseq so that all accesses are
locked and threads are not racing to get oseq values that they did not
claim.

This also fixes a build error in res_pktccops under dev mode.

(closes issue ASTERISK-23100)
Reported by: adomjan
Patch by: adomjan
........

Merged revisions 406037 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 406038 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 406049 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406078 65c4cc65-6c06-0410-ace0-fbb531ad65f3