........

Merged revisions 401096 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401097 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This makes it clear that the ARI API calls for listing channels and
bridges will list all channels or bridges in the system and not just
those that are in or are controlled by a Stasis application.

(closes issue ASTERISK-22635)
Reported by: Kevin Harwell
........

Merged revisions 401087 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401088 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When using realtime queues, queues have to be fetched from the database
every now and then to see if any info has been changed or to see if the
queue has been removed. When fetching info for an individual queue, the
pruning of other queues is unnecessarily costly.

Review: https://reviewboard.asterisk.org/r/2907/
........

Merged revisions 401049 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 401076 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 401077 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401079 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/2911/
........

Merged revisions 401040 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401041 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Consistently compare format2index() return value so matrix_get() cannot
get passed negative values.

* Optimize ast_translator_best_choice() to defer initializing things until
needed.  Also cached the matrix_get() return value rather than repeatedly
calling it.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401039 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 401030 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401031 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 401016 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 401017 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401020 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This adds the list of expected errors to the /bridges/{bridgeId}/record
ARI documentation so that outbound 4xx errors validate properly.
Previously, this would result in a response validation failure.

(closes issue ASTERISK-22627)
Reported by: Joshua Colp
........

Merged revisions 401018 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401019 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change updates how we handle toggle events, rather then create two
different function names, we'll just use POST / DELETE from HTTP to handle it.

Review: https://reviewboard.asterisk.org/r/2906/
........

Merged revisions 400999 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401007 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When a 200 OK for an initial INVITE is received, we were doing
the right thing by ACKing and sending an immediate BYE. However,
we also were doing the wrong thing and queuing an answer frame,
thus causing the call to be answered. This would cause the call
to be hung up by the channel thread, thus resulting in a second
BYE being sent out.

In this fix, I also have set the hangupcause to be correct since
the initial BYE being sent by Asterisk had an unknown hangup
cause. I have changed to using "Bearer capabilty not available"
since the call was hung up due to an SDP offer/answer error.

(closes issue ASTERISK-22621)
reported by Kinsey Moore
........

Merged revisions 400970 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 400971 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 400984 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400998 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Because I added a wiki_description to models and not their properties, the
rendered wiki page had the model description instead of the property
descriptions, which looks very silly indeed.

(closes issue ASTERISK-22705)
........

Merged revisions 400958 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400959 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Add hwtxgain and hwrxgain config options to chan_dahdi.conf with
documentation in chan_dahdi.conf.sample.

(closes issue ASTERISK-22429)
Reported by: Jaco Kroon
Patches:
      jira_asterisk_22429_hwgain_trunk.patch (license #5621) patch uploaded by rmudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400950 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Remember the swgain setting from CLI "dahdi set swgain" command so the
CLI "dahdi show channel" output will reflect the current setting.

* Updated CLI "dahdi set hwgain" and "dahdi set swgain" documentation.

(issue ASTERISK-22429)
Reported by: Jaco Kroon
Patches:
      jira_asterisk_22429_v1.8_v2.patch (license #5621) patch uploaded by rmudgett
........

Merged revisions 400907 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 400909 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 400911 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400913 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Bumping the SDP version number can cause interoperability problems
since receivers of the responses will expect that a 200 SDP will
be identical to a previous 183 SDP.

(closes issue ASTERISK-21204)
reported by NITESH BANSAL

Patches:
	dont-increment-session-version-in-2xx-after-183.patch uploaded by NITESH BANSAL (License #6418)
........

Merged revisions 400906 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 400908 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 400910 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400912 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If the server didn't exist that we are trying to register to the log message
would say that a 408 was received from that server when in reality one wasn't.
Added log messages stating no response was received if the response does not
exist.

(closes issue ASTERISK-22554)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2893/
........

Merged revisions 400890 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400891 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The module info block was repeated twice. Once is sufficient.
........

Merged revisions 400881 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400882 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The INVITE session state callback wrongly assumes that a session will always exist, but
when rapidly terminating the session this assumption goes out the window. As all handler
code for the INVITE session state callback requires the session it will now just exit
immediately if no session exists.

(closes issue ASTERISK-22668)
Reported by: John Bigelow
........

Merged revisions 400872 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400873 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When generating the list of authentication credentials to pass to
PJSIP, Asterisk was using the raw pointer of a pj_str_t which is not
always NULL-terminated. This sometimes resulted in incorrect text for
the realm and a failure to match the realm for authentication purposes
which was causing the outbound nominal auth pjsip basic call test to
bounce. This now uses the pj_str_t that contains the realm instead of
generating a new one. Thanks to John Bigelow for helping to narrow this
down.
........

Merged revisions 400863 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400864 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 400854 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400855 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
  r400508 | dlee | 2013-10-03 23:54:51 -0500 (Thu, 03 Oct 2013) | 1 line
  
  Corrected response class for stopPlayback
........
  r400842 | dlee | 2013-10-10 14:23:24 -0500 (Thu, 10 Oct 2013) | 1 line
  
  Correct some ARI wiki rendering errors
........
  r400843 | dlee | 2013-10-10 14:26:19 -0500 (Thu, 10 Oct 2013) | 1 line
  
  Updated /play resource docs. The playback of http: resources isn't implemented... yet
........
  r400848 | dlee | 2013-10-11 11:18:46 -0500 (Fri, 11 Oct 2013) | 5 lines
  
  Fix a stupid copy/paste error in ARI docs.
  
  Patches:
      ari-doc-patch.txt uploaded by jbigelow (license 5091)
........

Merged revisions 400508,400842-400843,400848 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400852 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400851 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The crash is caused by a race condition when switching between native RTP
and softmix bridging technologies.  In this situation, the bridging
technology is switched from native RTP to softmix, and then back to native
RTP fast enough that the softmix private data gets destroyed before the
softmix mixing thread gets started.

Thanks to Kinsey Moore for the crash analysis.

* Fix race condition when starting the softmix mixing thread and switching
to another bridge technology.

(closes issue ASTERISK-22678)
Reported by: John Bigelow
Patches:
      jira_asterisk_22678_v12.patch (license #5621) patch uploaded by rmudgett
Tested by: John Bigelow
........

Merged revisions 400849 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400850 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 400833 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400834 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change fixes two issues when setting an outbound proxy:

1. The outbound proxy URI was not parsed and validated during configuration.
2. If an outgoing dialog was created and the outbound proxy could not be set an assertion would
occur because the usage count on the dialog was not decremented.

The documentation has also been updated to specify that a full URI must be specified for
the outbound proxy.

(closes issue ASTERISK-22672)
Reported by: Antti Yrjola
........

Merged revisions 400824 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400825 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Using 'lu' will produce a compiler warning for some versions of gcc and on some
architectures. 'z' should be portable as a format specifier for size_t.
........

Merged revisions 400812 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400813 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds support to the PJSIP stack in Asterisk for SIP header
manipulation. Note that this is analagous to SIPAddHeader/SIPRemoveHeader.

For PJSIP_HEADER, an incoming supplemental session callback is registered that
takes the pjsip_hdrs from the incoming session and stores them in a linked
list in the session datastore.  Calls to PJSIP_HEADER traverse over the list
and return the nth matching header where 'n' is the 'number' argument to the
function.

When adding a header, the first call creates a datastore and linked list and
adds the datastore to the session.  The header is then created as a pjsip_hdr
and added to the list.  An outgoing supplemental session callback then
traverses the list and adds the headers to the outgoing pjsip_msg.

When removing a header, the list created with PJSIP_HEADER(add,...) is
traversed and all matching entries are removed.

(closes issue ASTERISK-22498)
Reported by: George Joseph
patch:
  res_pjsip_header_funcs_v1.patch uploaded by george.joseph (License 6322)
........

Merged revisions 400771 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400772 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When running configure, libiodbc2 development headers will fulfill the
requirement for ODBC development headers, but will not function
properly. This adds a warning when libiodbc2 development headers are
detected instead of unixodbc development headers.

(closes issue ASTERISK-22459)
Reported by: Patrick Maille
Tested by: Walter Doekes
Patches:
    issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes (License 5674)
........

Merged revisions 400767 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 400768 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 400769 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400770 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Clear the deferred_logoff flag when an agent logs in.

(closes issue ASTERISK-22669)
Reported by: John Bigelow
........

Merged revisions 400754 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400759 65c4cc65-6c06-0410-ace0-fbb531ad65f3

pjsip_strerror is only aware of PJSIP-specific error
codes. pj_strerror() is aware of all PJProject error
codes and OS-specific error codes.

This specifically fixes an oft-seen error in transport
configuration code where EADDRINUSE would result in
"Unknown PJSIP error 120098" instead of a useful
message.
........

Merged revisions 400749 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400750 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ConfBridge now has the ability to set the language of announcements to the
conference.  The language can be set on a bridge profile in
confbridge.conf or by the dialplan function
CONFBRIDGE(bridge,language)=en.

(closes issue ASTERISK-19983)
Reported by: Jonathan White
Patches:
      M19983_rev2.diff (license #5138) patch uploaded by junky (modified)
Tested by: rmudgett
........

Merged revisions 400741 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 400742 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400744 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fixed looking in the wrong profiles container to see if the default_user
profile is already created in verify_default_profiles().  The bridge
profile container is never going to hold user profiles. :)
........

Merged revisions 400723 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 400724 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400728 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The AST_CONFIG dialplan function defined in func_config.c allocates its
config file list entries using ast_malloc. List entry allocations
destined for use with Asterisk's linked list API must be ast_calloc()d
or otherwise initialized so that list pointers are set to NULL. These
uses of ast_malloc have been replaced by ast_calloc to prevent
dereferencing of uninitialized pointer values when traversing the list.

(closes issue ASTERISK-22483)
Reported by: Brian Scott
........

Merged revisions 400694 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 400697 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 400701 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400704 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Ensure that when chan_sip binds to the IPv6 any address ([::]), IPv4
candidates are also added.

(closes issue ASTERISK-21917)
Reported by: Torrey Searle
Patches:
    0023_ipv6_stun_crash.patch uploaded by Torrey Searle (License 5334)
........

Merged revisions 400681 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 400682 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400684 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If you run Asterisk in the background and then connect to
it through a separate console, the thread that runs CLI commands
is not registered with PJLIB. Thus PJLIB does not like it when
you attempt to send OPTIONS requests from that thread. So now
we push the task into the threadpool, which we know to be registered
with PJLIB.

Thanks to Antti Yrjola for reporting this.
........

Merged revisions 400680 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400683 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The https://reviewboard.asterisk.org/r/2888/ review changes manager to not
subscribe to stasis when it is disabled for performance reasons.  When
manager is disabled app_queue and res_agi decline to load and fail to
clean up what they have already allocated.

* Made app_queue and res_agi clean up allocated resources when they
decline to load.

* Made app_queue and res_agi use their own subscriptions to the stasis
topics instead of borrowing manager's message router structure
inappropriately.

(closes issue ASTERISK-22604)
Reported by: rmudgett

Review: https://reviewboard.asterisk.org/r/2902/
........

Merged revisions 400671 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400672 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 400661 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400662 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Commit r62462 added two extra fields for logging "the original position the
caller entered the queue at, and the amount of time the caller was waiting in
the queue."  But when r75969 was merged from 1.4 into trunk (r75977), these two
fields disappeared. Those two extra fields were not logged in 1.4 and when the
patch was merged, those fields went away.

Therefore, this is a regression and was caught by the reporter because he was
reading the awesome "Asterisk: The Definitive Guide" book.

(closes issue ASTERISK-22197)
Reported by: Dalius M.
Tested by: Dalius M.
Patches:
    asterisk-22197-q-log-exitwithkey.diff
				     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2901/
........

Merged revisions 400622 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 400623 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 400624 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400625 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 400588 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400593 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds IPv6 support to chan_iax2.  Yay!

(closes issue ASTERISK-22025)
Patches:
  iax2-ipv6-v5-reviewboard.diff by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2660/
........

Merged revisions 400567 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400568 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........

Merged revisions 400552 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400553 65c4cc65-6c06-0410-ace0-fbb531ad65f3