(issue ASTERISK-21800)



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@390229 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-21800)
Reported by: Dmitry Melekhov
Patches:
        ASTERISK-21800-1.patch
Tested by: Dmitry Melekhov
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@390228 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Check the returned bridged pointer for NULL to avoid a crash.  It looks
like chan_agent is returning a NULL pointer when it probably should be
returning a pointer to the channel the Agent channel is pretending to be.

(closes issue ASTERISK-21793)
Reported by: Rodrigo P. Telles
Patches:
      jira_asterisk_21793_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Rodrigo P. Telles
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Merged revisions 390044 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@390047 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Reported by: Michael Walton
Tested by: Jonathan Rose
Patches:
    slinfactory.c.ASTERISK-21799.patch uploaded by Michael Walton (license 6502)
(closes issue ASTERISK-21799)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@389896 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When Asterisk shuts down and shuts down the loggin gsubsystem, any
messages currently in flight will not get logged. This patch prevents the
loop writing messages from breaking out prematurely, such that all of the
messages are logged.

(closes issue ASTERISK-21716)
Reported by: Corey Farrell
patches:
  logger-process-all-messages.patch uploaded by Corey Farrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@389677 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix several problems caused by multiple line usage with i2004 phones.
Reported by: Daniel Bohling, MihaiMircea

(closes issue ASTERISK-21061)
(closes issue ASTERISK-21120)



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@389661 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@389245 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If DEBUG_THREADS is enabled __ast_rwlock_destroy causes a segfault while trying
to access a possible NULL t->track object.  A NULL check has been added before
trying to access the memory.

(closes issue ASTERISK-21724)
Reported by: Corey Farrell
Fixed by: Corey Farrell
Patches:
	ast_rwlock_destroy-segv.patch uploaded by Corey Farrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388839 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The snapshot API contains an option that allow for combining of new 
and old messages within a single snapshot. New messages, however, 
include options beyond just 'INBOX' - it also includes the Urgent 
folder. A previous patch that combined INBOX and Urgent accidentally 
impacted snapshots that attempted to gain messages from just the Old 
folder. This patch fixes the snapshot gathering such that the API 
returns the appropriate messages for the folder selected, with and 
without the combine option.

This should make it more clear about what's happening.

Review: https://reviewboard.asterisk.org/r/2539/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388816 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This allows the SRTP library to be shut down properly when the
functionality is offered by libsrtp.

Review: https://reviewboard.asterisk.org/r/2538/
(closes issue ASTERISK-21719)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388769 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The debug versions of ao2_ref() should only be used if REF_DEBUG is
enabled so nothing is written to /tmp/refs unexpectedly.

(closes issue ASTERISK-21785)
Reported by: abelbeck
Patches:
      jira_asterisk_21785_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: abelbeck


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388700 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The CALL-ID (ie [C-00000074]) is missing when logging to syslog.  This was just
an oversight when this feature was added.

* Add CALL-IDs when using syslog

(closes issue ASTERISK-21430)
Reported by: Nikola Ciprich
Tested by: Nikola Ciprich, Michael L. Young
Patches:
    asterisk-21430-syslog-callid_trunk.diff by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2526/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388605 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The prior code committed, r385473, failed to take into consideration that not
all outgoing calls will be to a peer.  My fault.

This patch does the following:

* Check if there is a related peer involved.  If there is, check and set NAT 
  settings according to the peer's settings.

* Fix a problem with realtime peers.  If the global setting has auto_force_rport
  set and we issued a "sip reload" while a peer is still registered, the peer's
  flags for NAT are reset to off.  When this happens, we were always setting the
  contact address of the peer to that of the full contact info that we had.

(closes issue ASTERISK-21374)
Reported by: jmls
Tested by: Michael L. Young
Patches:
   asterisk-21374-fix-crash-and-rt-peers.diff by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2524/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388601 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adding the cleanup function needs some deeper thought since it
apparently doesn't exist for all variants of libsrtp.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388597 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-21723)
Reported by: Corey Farrell
Patches:
    core-pbx-cleanup.patch uploaded by Correy Farrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388578 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Ensure that libsrtp is shutdown properly when res_srtp is unloaded.

(closes issue ASTERISK-21719)
Reported by: Corey Farrell
Patches:
    res_srtp-library-shutdown.patch uploaded by Corey Farrell
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388530 65c4cc65-6c06-0410-ace0-fbb531ad65f3

AMI actions must never return non-zero unless they intend to close the AMI
connection.  (Which is almost never.)

(closes issue ASTERISK-21779)
Reported by: Paul Goldbaum
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388478 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made isdn_msg_parser.c build a progress message with the mandatory
progress indicator IE.  (The mISDNuser NT state machine rejected sending
the incomplete message.)

Note: The associated mISDN and mISDNuser patches respectively are viewable
here:
http://svnview.digium.com/svn/thirdparty?view=rev&rev=200
http://svnview.digium.com/svn/thirdparty?view=rev&rev=201

(closes issue AST-1153)
Reported by: Guenther Kelleter
Patches:
      progress-chan_misdn.diff (license #6372) patch uploaded by Guenther Kelleter
      progress-misdn.diff (license #6372) mISDN patch uploaded by Guenther Kelleter
      progress-misdnuser.diff (license #6372) mISDNuser patch uploaded by Guenther Kelleter
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388424 65c4cc65-6c06-0410-ace0-fbb531ad65f3

pbx_dundi added an io context without removing
it. This caused a memory leak when the module was
unloaded.

(closes ASTERISK-21718)
Reported by Corey Farrell
Patches:
	pbx_dundi-ast_io_remove.patch uploaded by Corey Farrell (License #5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388253 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When we send out a CN packet (for instance, in the case of using rtpkeepalives),
we are not setting the payload code properly.  Also, we are setting the marker
bit when we shouldn't be according to RFC 3389, section 4.

AST_RTP_CN is not defined by AST_FORMAT codes.  Therefore, we should be using
ast_rtp_codecs_payload_code() rather than ast_rtp_codecs_payload_lookup().

11 and trunk already use the appropriate function.

* In 1.8, use ast_rtp_codecs_payload_code()

* Remove the setting of the marker bit

* Fix the debug message by incrementing the seqno after the debug message is set
  in order to display the correct seqno that was sent out

(closes issue ASTERISK-21246)
Reported by: Peter Katzmann
Tested by: Peter Katzmann, Michael L. Young
Patches:
    asterisk-21246-rtp-cng-payload-error_1.8_v2.diff
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2500/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388112 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When the "ignorebusy" setting was deprecated, we added some code to allow us to
be compatible with older setups that are still using the "ignorebusy" setting
instead of "ringinuse".  We set a char *variable with the column name to use,
which helps the realtime functions to use the correct column in their SQL
queries.  When "persistentmembers" is enabled, we are not setting this variable
before the realtime functions were called to load members.  This results in the
variable being NULL and therefore causing a segfault when loading members during
the module's process of loading.

The solution was to move the code that sets that variable to be before these
realtime functions are called during the loading of the module.

(closes issue ASTERISK-21738)
Reported by: JoshE
Tested by: JoshE
Patches:
    asterisk-21738-rt-ringinuse-field-not-set.diff
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2499/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388108 65c4cc65-6c06-0410-ace0-fbb531ad65f3

RFC6665 4.2.2: ... after a failed State NOTIFY transaction remove the subscription

The problem is that the State Notify requests rely on the 200OK reponse for pacing control
and to not confuse the notify susbsystem.
The issue is, the pendinginvite isn't cleared if a response isn't received,
thus further notify's are never sent.

The solution, follow RFC 6665 4.2.2's 'SHOULD' and remove the subscription after failure.
  
(closes issue ASTERISK-21677)

Reported by: Dan Martens
Tested by: Dan Martens, David Brillert, alecdavis
alecdavis (license 585)

Review https://reviewboard.asterisk.org/r/2475/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387880 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The \example tags marks an entire file as an example, not a code snippet.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387823 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Reload support was originally not included for SLA.  It was added later,
but in a fairly non-traditional way.  It basically sets a flag
indicating that a reload is pending, and then waits for a time where it
thinks everything SLA related is idle and unused, and *then* executes
the reload.  It does this because the reload process is destructive.  It
starts by throwing everything away and starting over.

There are a number of problems with this approach.  One of them is that
the check to see if anything in use was incomplete.  This patch makes it
more complete and thus less likely for a crash to occur during reload
processing.  However, this approach still has problems so some much more
significant reworking of this code will need to come in as a next step.

Patch credit and testing by CoreDial, LLC.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387689 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Alec's patch that added the Asterisk version to 'core show locks' angered the
items in utils, as they exist somewhat outside of the Asterisk build system.
Some day, this Makefile should get nuked from high orbit, but for now, include
version.c in its list of stuff to pile in.
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chan_sip: Session-Expires: Set timer to correctly expire at (~2/3) of the interval when not the refresher

RFC 4028 Section 10
	if the side not performing refreshes does not receive a
	session refresh request before the session expiration, it SHOULD send
	a BYE to terminate the session, slightly before the session
	expiration.  The minimum of 32 seconds and one third of the session
	interval is RECOMMENDED.

Prior to this asterisk would refresh at 1/2 the Session-Expires interval,
or if the remote device was the refresher, asterisk would timeout at interval end.

Now, when not refresher, timeout as per RFC noted above.

(closes issue ASTERISK-21742)

Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)

Review https://reviewboard.asterisk.org/r/2488/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387345 65c4cc65-6c06-0410-ace0-fbb531ad65f3

chan_sip: Honor Session-Expires in 200OK response when it's a RE-INVITE when asterisk is the refresher.

RFC 4028 Section 7.2
 "UACs MUST be prepared to receive a Session-Expires header field in a
 response, even if none were present in the request." 

What changed
  After ASTERISK-20787, inbound calls to asterisk with no Session-Expires in the INVITE are now are offered
  a Session-Expires (1800 asterisk default) in the response, with asterisk as the refresher.

Symptom:
  After 900 seconds (asterisk default refresher period 1800), asterisk RE-INVITEs the device, the device
   may respond with a much lower Session-Expires (180 in our case) value that it is now using.

  Asterisk ignores this response, as it's deemed both an INBOUND CALL, and a RE-INVITE.

  After 180 seconds the device times out and sends BYE (hangs up), asterisk is still working with the
  refresher period of 1800 as it ignored the 'Session Expires: 180' in the previous 200OK response.
 
Fix:
	handle_response_invite() when 200OK, remove check for outbound and reinvite.
  
(closes issue ASTERISK-21664)

Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)

Review https://reviewboard.asterisk.org/r/2463/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387319 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Lower bound of a 16bit signed int is -32768 not -32767

(closes issue ASTERISK-21744)

Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387298 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Assist with reporting 'core show locks' when submitting bug reports.

Example below:

===========================
== SVN-branch-1.8-...
== Currently Held Locks
===========================


(closes issue ASTERISK-21743)

Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387295 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In certain situations, when the RTP engine goes to send a DTMF end digit
it may be in a situation where the remote address is no longer available,
or the digit that was supposed to be sent is invalid. In such cases, we
need to clear the RTP counters appropriately. Otherwise, when the RTP
source is set again, we'll continue to think that we're in the middle of
sending a DTMF digit, which can confuse the remote party (signficantly).

(closes issue ASTERISK-21522)
Reported by: Corey Farrell
patches:
  rtp_dtmf_process_end.patch uploaded by Corey Farrell (License 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387216 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When you have lots of SIP peers (according to the issue reporter, around 3500),
the 'sip show peers' CLI command or AMI action can crash due to a poorly placed
string duplication that occurs on the stack. This patch refactors the command
to not allocate the string on the stack, and handles the formatting of a single
peer in a separate function call.

(closes issue ASTERISK-21466)
Reported by: Guillaume Knispel
patches:
  fix_sip_show_peers_stack_overflow_asterisk_11.3.0-v2.patch uploaded by gknispel (License 6492)



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387134 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Way back when in the dark days of Asterisk 1.8.9, blind transferring a call
in a context that included the 'h' extension would inadvertently execute the
hangup code logic on the transferred channel. This was a "bad thing". The fix
was to properly check for the softhangup flags on the channel and only execute
the 'h' extension logic (and, in later versions, hangup handler logic) if the
channel was well and truly dead (Jim).

Unfortunately, CDRs are fickle. Setting the softhangup flag when we detected
that the channel was leaving the bridge (but not to die) caused some crucial
snippet of CDR code, lying in ambush in the middle of the bridging code, to
not get executed. This had the effect of blowing away one of the CDRs that is
typically created during a blind transfer.

While we live and die by the adage "don't touch CDRs in release branches", this
was our bad. The attached patch restores the CDR behavior, and still manages to
not run the 'h' extension during a blind transfer (at least not when it's
supposed to).

Thanks to Steve Davies for diagnosing this and providing a fix.

Review: https://reviewboard.asterisk.org/r/2476

(closes issue ASTERISK-21394)
Reported by: Ishfaq Malik
Tested by: Ishfaq Malik, mjordan
patches:
  fix_missing_blindXfer_cdr2 uploaded by one47 (License 5012)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387030 65c4cc65-6c06-0410-ace0-fbb531ad65f3

16 bit signed integers have a range of [-32768, 32768).  The existing code
was using the interval (-32768, 32768) instead.  This patch fixes that.

Review: https://reviewboard.asterisk.org/r/2479/
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1.4.24 core sounds includes a full set of Italian prompts for core sounds and a fix for the missing voicemail prompts in the Russian language.

(closes issue ASTERISK-19431)
(closes issue ASTERISK-19721)
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Review: https://reviewboard.asterisk.org/r/2263/
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If a system allows for its usage, Asterisk will use glob to help parse
Asterisk .conf files. The config file loading routine was leaking the memory
allocated by the glob() routine when the config file was in an unmodified
or invalid state.

This patch properly calls globfree in those off nominal paths.

(closes issue ASTERISK-21412)
Reported by: Corey Farrell
patches:
  config_glob_leak.patch uploaded by Corey Farrell (license 5909)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@386677 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch cleans up two things features:
* It properly unregisters the CLI commands that features registered
* It cancels and performs a pthread_join on the created parking thread. This
  not only properly joins a non-detached thread, but also prevents disposing
  of the parking lots prior to the parking thread completely exiting.

(closes issue ASTERISK-21407)
Reported by: Corey Farrell
patches:
  features_shutdown-r2.patch uploaded by Corey Farrell (License 5909)
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