asterisk/asterisk.git
5 years agochan_sip: Clarify 'Forcerport' Setting Displayed When Running "sip show peers"
Michael L. Young [Mon, 28 Oct 2013 14:59:16 +0000 (14:59 +0000)]
chan_sip: Clarify 'Forcerport' Setting Displayed When Running "sip show peers"

While looking at ASTERISK-22236, Walter Doekes pointed out that when running
"sip show peers", the setting being displayed can be confusing.  The display of
"N" used to mean NAT (i.e. yes).  The NAT setting has gone through many
different changes resulting in the display of different characters to try and
convey what the current setting is for 'Forcerport' (A for Auto and Forcerport
is currently on, a for Auto but Forcerport is off, Y for yes, and N for no).
During the initial code review to try and clarify these settings (especially
since "N" no longer meant what it used to mean in prior versions of Asterisk),
Mark Michelson suggested using the full space available to display the settings
which helped to make the settings very clear.  That was a great suggestion.

Therefore, this patch does the following:

* The column for 'Forcerport' now will show: Auto (Yes), Auto (No), Yes, or No.

* A column for the 'Comedia' setting has been added.  It too will display the
  setting in a non-cryptic way: Auto (Yes), Auto (No), Yes, or No.

* UPGRADE.txt has been updated to document this change.

(closes issue ASTERISK-22728)
Reported by: Walter Doekes
Tested by: Michael L. Young
Patches:
    asterisk-forcerport-display-clarification_v3.diff
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2941
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Merged revisions 402111 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 402112 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402113 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoFilter out internal channels from dial message handling
Matthew Jordan [Sun, 27 Oct 2013 23:22:51 +0000 (23:22 +0000)]
Filter out internal channels from dial message handling

Surrogate channels would pop up from time to time in dial message handling.
This would cause a WARNING message to appear, indicating that the Surrogate
channel had no CDR. This patch filters out those channels that have the
internal implementation flag set, such that the WARNING message isn't
displayed.
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Merged revisions 402090 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402091 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoPrevent CDR backends from unregistering while billing data is in flight
Matthew Jordan [Sun, 27 Oct 2013 20:04:17 +0000 (20:04 +0000)]
Prevent CDR backends from unregistering while billing data is in flight

This patch makes it so that CDR backends cannot be unregistered while active
CDR records exist. This helps to prevent billing data from being lost during
restarts and shutdowns.

Review: https://reviewboard.asterisk.org/r/2880/
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Merged revisions 402081 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402082 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoUpdate Alembic database scripts for external scripting and PostgreSQL, Oracle
Matthew Jordan [Sun, 27 Oct 2013 02:39:34 +0000 (02:39 +0000)]
Update Alembic database scripts for external scripting and PostgreSQL, Oracle

This patch does the following:
1) The env scripts have been updated to be tolerant of a NULL configuration
   file. This occurs when configuration is provided by an external script,
   such that the actual config.ini file is not used.
2) Enum types have all been given names. This is needed for PostgreSQL script
   generation.
3) The identifier meetme_confno_starttime_endtime is greater than 30
   characters, and hence invalid for Oracle databases. This has been truncated
   down to meetme_confno_start_end.
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Merged revisions 400383 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402073 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agochan_pjsip: Fix a crash when direct media is enabled and an ACK is received after...
Joshua Colp [Sat, 26 Oct 2013 12:56:08 +0000 (12:56 +0000)]
chan_pjsip: Fix a crash when direct media is enabled and an ACK is received after the channel is hung up.

(closes issue ASTERISK-22731)
Reported by: Kinsey Moore
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Merged revisions 402064 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402065 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_stasis.c: Made use the ao2_container callback templates.
Richard Mudgett [Sat, 26 Oct 2013 00:36:31 +0000 (00:36 +0000)]
res_stasis.c: Made use the ao2_container callback templates.

* Made res_stasis.c use the OBJ_SEARCH_XXX defines.
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Merged revisions 402055 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402056 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agortp_engine: fix rtp payloads copy and improve argument names
Scott Griepentrog [Sat, 26 Oct 2013 00:27:02 +0000 (00:27 +0000)]
rtp_engine: fix rtp payloads copy and improve argument names

In function ast_rtp_instance_early _bridge_make_compatible the
use of instance 0/1 as arguments doesn't clearly communicate a
direction that the copying of payloads from the source channel
to the destination channel will occur, making it more probable
to have the arguments to ast_rtp_codecs_payloads_copy() put in
the reverse order.  This patch renames the arguments with _dst
and _src suffixes and corrects the copy direction.

(closes issue ASTERISK-21464)
Reported by: Kevin Stewart
Review: https://reviewboard.asterisk.org/r/2894/
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Merged revisions 402000 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Test shows rtpmap:119 being copied per this change, but is not in sip invite
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Merged revisions 402042 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 402043 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402054 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agotaskprocessor: Made use pthread_equal() to compare thread ids.
Richard Mudgett [Fri, 25 Oct 2013 23:58:32 +0000 (23:58 +0000)]
taskprocessor: Made use pthread_equal() to compare thread ids.

* Removed another silly use of RAII_VAR().  RAII_VAR() and SCOPED_LOCK()
are not silver bullets that allow you to turn off your brain.
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Merged revisions 402044 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402045 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoYou'd think that new files would be free of whitespace issues. But you would be...
Richard Mudgett [Fri, 25 Oct 2013 22:03:04 +0000 (22:03 +0000)]
You'd think that new files would be free of whitespace issues.  But you would be wrong.
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Merged revisions 402003 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402004 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoARI: channel/bridge recording errors when invalid format specified
Jonathan Rose [Fri, 25 Oct 2013 22:01:43 +0000 (22:01 +0000)]
ARI: channel/bridge recording errors when invalid format specified

Asterisk will now issue 422 if recording is requested against channels
or bridges with an unknown format

(closes issue ASTERISK-22626)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2939/
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Merged revisions 402001 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402002 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoARI recordings: Issue HTTP failures for recording requests with file conflicts
Jonathan Rose [Fri, 25 Oct 2013 21:28:32 +0000 (21:28 +0000)]
ARI recordings: Issue HTTP failures for recording requests with file conflicts

If a file already exists in the recordings directory with the same name as what
we would record, issue a 422 instead of relying on the internal failure and
issuing success.

(closes issue ASTERISK-22623)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2922/
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Merged revisions 401973 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401999 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agopbx.c: fix confused match caller id that deleted exten still in hash
Scott Griepentrog [Fri, 25 Oct 2013 20:51:13 +0000 (20:51 +0000)]
pbx.c: fix confused match caller id that deleted exten still in hash

This fixes a bug where a zero length callerid match adjacent to a no
match callerid extension entry would be deleted together, which then
resulted in hashtable references to free'd memory.  A third state of
the matchcid value has been added to indicate match to any extension
which allows enforcing comparison of matchcid on/off without errors.

(closes issue AST-1235)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/2930/
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Merged revisions 401959 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401960 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401961 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401962 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoPJSIP: Add log messages when requests are received for non-existent endpoints
Jonathan Rose [Fri, 25 Oct 2013 17:41:38 +0000 (17:41 +0000)]
PJSIP: Add log messages when requests are received for non-existent endpoints

(closes issue ASTERISK-22552)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2934/
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Merged revisions 401938 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401939 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoPut clicompat-r2.patch back in
Jonathan Rose [Fri, 25 Oct 2013 17:32:17 +0000 (17:32 +0000)]
Put clicompat-r2.patch back in

We've figured out how to resolve the problems this was causing in 12/trunk,
so this can go back in now.

(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
    clicompat-r2.patch uploaded by coreyfarrell (license 5909)
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Merged revisions 401914 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401935 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401936 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401937 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agorevert clicompat-r2.patch from r401704
Jonathan Rose [Fri, 25 Oct 2013 16:59:33 +0000 (16:59 +0000)]
revert clicompat-r2.patch from r401704

Patch caused the following build errors against testsuite
https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD4-244

(issue ASTERISK-22467)
Reported by: Corey Farrell
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Merged revisions 401895 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401896 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401897 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401898 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agochan_sip: Allow a sip peer to accept both AVP and AVPF calls
Kevin Harwell [Fri, 25 Oct 2013 16:09:05 +0000 (16:09 +0000)]
chan_sip: Allow a sip peer to accept both AVP and AVPF calls

Adapts the behaviour of avpf to only impact the format of outgoing calls. For
inbound calls, both AVP and AVPF calls will be accepted regardless of the value
of avpf in the configuration.

(closes issue ASTERISK-22005)
Reported by: Torrey Searle
Patches:
     optional_avpf_trunk.patch uploaded by tsearle (license 5334)
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Merged revisions 401884 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401885 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401886 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoBlocked revisions 401391
David M. Lee [Fri, 25 Oct 2013 13:50:57 +0000 (13:50 +0000)]
Blocked revisions 401391

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Blocked revisions 401379

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chan_dahdi: Fix unable to get index warning when transferring an analog call.

Transferring an analog call using flashhooks generated an unable to get
index WARNING message when the transfer is completed.

* Removed unnecessary analog subchannel shell games when transferring a
call using flashhooks.

Thanks to Tzafrir Cohen for mentioning this in a comment on issue
ASTERISK-22720.
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Merged revisions 401378 from http://svn.asterisk.org/svn/asterisk/branches/1.8

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401874 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agotest_json: Fix deprecation warnings
David M. Lee [Fri, 25 Oct 2013 13:49:20 +0000 (13:49 +0000)]
test_json: Fix deprecation warnings

After a series of upgrades over recent weeks, I've discovered that
test_json.c won't compile in dev mode any more for me.

One of gcc-4.8.2, OS X Mavericks or Xcode 5 has decided to deprecate
tempnam. Which, in general, is a good thing. But for test code that just
needs a temporary file, it's just annoying.

This patch replaces usage of tempname with mkstemp, avoiding the
deprecation warning. It also removes the temporary files when the test
is complete, which apparently we weren't doing before (oops).

Review: https://reviewboard.asterisk.org/r/2957/
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Merged revisions 401872 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401873 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoLogging: Logging types ignored after specifying a verbose level
Kevin Harwell [Thu, 24 Oct 2013 21:06:14 +0000 (21:06 +0000)]
Logging: Logging types ignored after specifying a verbose level

If one specified a verbose level within a logging facility in
logger.conf then any component after it was ignored.  Fixed so
all values are correctly read.

(closes issue ASTERISK-22456)
Reported by: Kevin Harwell
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Merged revisions 401833 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401835 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401836 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoThe Swagger 1.2 specification for type extension ended up being
David M. Lee [Thu, 24 Oct 2013 20:48:17 +0000 (20:48 +0000)]
The Swagger 1.2 specification for type extension ended up being
slightly different than my proposal. Instead of putting an 'extends'
field on the subtype, the base type has a 'subTypes' field, which is a
list of the subTypes. Given that its a messaging model and not an
object model, kinda makes sense.

This patch changes the events.json api-doc, and the python translators
to take the new format into account.

Other changes that are in Swagger 1.2 were not adopted, since the spec
is still in flux, and could change before it's finalized.

A summary of changes to the Swagger-1.2 spec can be found at
https://github.com/wordnik/swagger-core/wiki/1.2-transition.

(closes issue ASTERISK-22440)
Review: https://reviewboard.asterisk.org/r/2909/
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Merged revisions 401701 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401834 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoutils: Fix memory leaks and missed unregistration of CLI commands on shutdown
Jonathan Rose [Thu, 24 Oct 2013 20:34:53 +0000 (20:34 +0000)]
utils: Fix memory leaks and missed unregistration of CLI commands  on shutdown

Final set of patches in a series of memory leak/cleanup patches by Corey Farrell

(closes issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
    main-utils-1.8.patch uploaded by coreyfarrell (license 5909)
    main-utils-11.patch uploaded by coreyfarrell (license 5909)
    main-utils-12up.patch uploaded by coreyfarrell (license 5909)
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Merged revisions 401829 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401830 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401831 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401832 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agotest_linkedlists: Fix memory leak
Jonathan Rose [Thu, 24 Oct 2013 19:57:04 +0000 (19:57 +0000)]
test_linkedlists: Fix memory leak

(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
    test_linkedlists-1.8.patch uploaded by coreyfarrell (license 5909)
    test_linkedlists-11up.patch uploaded by coreyfarrell (license 5909)
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Merged revisions 401790 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401791 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401792 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401793 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agojitterbuf: Fix memory leak on jitter buffer reset
Jonathan Rose [Thu, 24 Oct 2013 19:42:21 +0000 (19:42 +0000)]
jitterbuf: Fix memory leak on jitter buffer reset

(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
    jitterbuf-jb_reset-leak-1.8.patch
    jitterbuf-jb_reset-leak-11up.patch
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Merged revisions 401786 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401787 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401788 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401789 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoastobj2: Unregister debug CLI commands at exit
Jonathan Rose [Thu, 24 Oct 2013 19:31:23 +0000 (19:31 +0000)]
astobj2: Unregister debug CLI commands at exit

(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
    astobj2-clean-debug-cli-1.8-11.patch uploaded by coreyfarrell (license 5909)
    astobj2-clean-debug-cli-12up.patch uploaded by coreyfarrell (license 5909)
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Merged revisions 401781 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401783 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401784 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401785 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoapp_voicemail: Memory Leaks against tests
Jonathan Rose [Thu, 24 Oct 2013 18:46:56 +0000 (18:46 +0000)]
app_voicemail: Memory Leaks against tests

(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
    app_voicemail-1.8.patch uploaded by coreyfarrell (license 5909)
    app_voicemail-11up.patch uploaded by coreyfarrell (license 5909)
........

Merged revisions 401743 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401744 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401745 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401746 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agomemory leaks: Memory leak cleanup patch by Corey Farrell (second set)
Jonathan Rose [Thu, 24 Oct 2013 17:00:27 +0000 (17:00 +0000)]
memory leaks: Memory leak cleanup patch by Corey Farrell (second set)

Also covers ast_app_parse_timelen-fail-zero-length.patch, but the patch was
replaced with one of my own.

(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
    chan_dahdi-cleanup_push.patch uploaded by coreyfarrell (license 5909)
    clicompat-r2.patch uploaded by coreyfarrell (license 5909)
    codecs-ilbc-doCPLC.patch uploaded by coreyfarrell (license 5909)
    data-cleanup-test-registration.patch uploaded by coreyfarrell (license 5909)
    main-asterisk-kill-listener.patch uploaded by coreyfarrell (license 5909)
........

Merged revisions 401704 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401705 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401706 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401707 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agomemory leaks: Memory leak cleanup patch by Corey Farrell (first set)
Jonathan Rose [Wed, 23 Oct 2013 20:10:30 +0000 (20:10 +0000)]
memory leaks: Memory leak cleanup patch by Corey Farrell (first set)

(issue ASTERSIK-22467)
Reported by: Corey Farrell
Patches:
    chan_sip-parse_contact_header_test-free-contacts.patch uploaded by coreyfarrell (license 5909)
    cli-filename-completion-leak.patch uploaded by coreyfarrell (license 5909)
    func_math.patch uploaded by corefarrell (license 5909)
    main-test-cleanup.patch uploaded by coreyfarrell (license 5909)
    test_dlinklists.patch uploaded by coreyfarrell (license 5909)
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Merged revisions 401660 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401661 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401662 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401663 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_rtp_asterisk: Address jittery DTMF events in RTP streams
Jonathan Rose [Wed, 23 Oct 2013 17:56:44 +0000 (17:56 +0000)]
res_rtp_asterisk: Address jittery DTMF events in RTP streams

(closes issue ASTERISK-21170)
Reported by: NITESH BANSAL
Patches:
    dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418)
Review: https://reviewboard.asterisk.org/r/2938/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401622 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agocdr_adaptive_odbc: Also apply a filter when the CDR value is empty.
Richard Mudgett [Wed, 23 Oct 2013 16:52:11 +0000 (16:52 +0000)]
cdr_adaptive_odbc: Also apply a filter when the CDR value is empty.

Extra CDR records are written if a filtered CDR value is empty because the
filter is not checked.

(closes issue ASTERISK-22272)
Reported by: Jordi Llull Chavarria
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401582 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoAdd a test suite event to indicate when the atxfer 3-way feature is detected
John Bigelow [Wed, 23 Oct 2013 16:48:39 +0000 (16:48 +0000)]
Add a test suite event to indicate when the atxfer 3-way feature is detected

This adds a test suite event that indicates to tests when the attended transfer
three-way call feature is detected.

Review: https://reviewboard.asterisk.org/r/2912/
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Merged revisions 401578 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401580 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agochan_mgcp: Properly handle malformed media lines
Kinsey Moore [Wed, 23 Oct 2013 15:23:58 +0000 (15:23 +0000)]
chan_mgcp: Properly handle malformed media lines

This corrects a situation in which a media line was not parsed properly
and resulted in a crash.

(closes issue ASTERISK-21190)
Reported by: adomjan
Patches:
    chan_mgcp.c-sscnaf_fix uploaded by adomjan (License 5448)
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Merged revisions 401537 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401538 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401539 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401540 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agochan_sip: Fix an issue where an incompatible audio format may be added to SDP.
Joshua Colp [Wed, 23 Oct 2013 11:16:44 +0000 (11:16 +0000)]
chan_sip: Fix an issue where an incompatible audio format may be added to SDP.

If preferred codecs included any non-audio format the code would
mistakenly add the audio format, even if it was not a joint capability
with the remote side.

(closes issue ASTERISK-21131)
Reported by: nbougues
Patches:
patch_unsupported_codec_1.8.patch uploaded by nbougues (license 6470)
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Merged revisions 401497 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401500 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agochan_iax2: Fix Binding To Multiple Addresses Again
Michael L. Young [Wed, 23 Oct 2013 02:36:01 +0000 (02:36 +0000)]
chan_iax2: Fix Binding To Multiple Addresses Again

When reworking chan_iax2 for IPv6, the ability to bind to multiple addresses
was removed by mistake.  This patch restores this functionality and adds notes
about IPv6 addresses in the sample config.

(closes issue ASTERISK-22741)
Reported by: Joshua Colp
Tested by: Michael L. Young
Patches:
    asterisk-22741-fix-binding-multiple-addr.diff
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2945/
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Merged revisions 401488 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401489 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_rtp_asterisk: Fix crash when RTCP is not available during SSRC change
Matthew Jordan [Tue, 22 Oct 2013 23:10:22 +0000 (23:10 +0000)]
res_rtp_asterisk: Fix crash when RTCP is not available during SSRC change

In r400089, a patch was put in to correct erroneous RTCP statistic resets.
Unfortunately, ast_rtp_read can be called on an RTP instance that does not
have RTCP information. This patch prevents that crash by only resetting
the statistics if we do actually have an RTCP instance.

(issue AST-1174)

(closes issue ASTERISK-22667)
Reported by: John Bigelow
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Merged revisions 401445 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401446 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401447 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401450 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoapp_queue: Fix CLI "queue remove member" queue_log entry.
Richard Mudgett [Tue, 22 Oct 2013 19:04:53 +0000 (19:04 +0000)]
app_queue: Fix CLI "queue remove member" queue_log entry.

The queue_log entry resulting from CLI "queue remove member" when
log_membername_as_agent is enabled is wrong.  It always uses the interface
name instead of the member name in the queue_log entry.

* Get the queue member before removing it from the queue so the member
name is available for the queue_log entry.

(closes issue ASTERISK-21826)
Reported by: Oscar Esteve
Patches:
      fix_membername.diff (license #6505) patch uploaded by Oscar Esteve
         (modified to fix potential ref leak)
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Merged revisions 401433 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401434 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401435 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoBridging: Fix orphaned bridge if neither of the joining channels can join.
Richard Mudgett [Tue, 22 Oct 2013 17:06:21 +0000 (17:06 +0000)]
Bridging: Fix orphaned bridge if neither of the joining channels can join.

The original issue noted that the bridge is orphaned when res_parking.so
is not loaded and a call uses the dial kK flags.

A similar issue happens when only one of the park flags is used.  In this
case you have the bridge with one or the other channel left in it.  The
channel and bridge will stay around until the channel hangs up.

* Fixed the initial bridge channel push failure to act as if the channel
were kicked out of the bridge.  The bridge then decides if it needs to be
dissolved.

(closes issue ASTERISK-22629)
Reported by: Kevin Harwell

Review: https://reviewboard.asterisk.org/r/2928/
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Merged revisions 401424 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401425 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_parking: Give parking timeout comebacktoorigin channel DTMF features.
Richard Mudgett [Tue, 22 Oct 2013 16:33:16 +0000 (16:33 +0000)]
res_parking: Give parking timeout comebacktoorigin channel DTMF features.

Parking timeouts did not set any DTMF features for the channel calling the
parker back.

* Added code to set the parkedcalltransfers, parkedcallreparking,
parkedcallhangup, and parkedcallrecording options appropriately for the
channels when a parking timeout occurs.  The recall channel DTMF options
are set using the BRIDGE_FEATURES channel variable to allow the other
timeout options to have the DTMF features available.

(closes issue ASTERISK-22630)
Reported by: Kevin Harwell

Review: https://reviewboard.asterisk.org/r/2942/
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Merged revisions 401422 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401423 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_parking: Update XML documention for DTMF features after parking timeout.
Richard Mudgett [Tue, 22 Oct 2013 16:28:05 +0000 (16:28 +0000)]
res_parking: Update XML documention for DTMF features after parking timeout.

* Updated the XML documentation to indicate that the parkedcalltransfers,
parkedcallreparking, parkedcallhangup, and parkedcallrecording
configuration options also apply to parking timeouts.

(issue ASTERISK-22630)
Reported by: Kevin Harwell

Review: https://reviewboard.asterisk.org/r/2942/
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Merged revisions 401420 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401421 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoAdd an 'R' option to Dial which sends ringing until early media has been received.
Joshua Colp [Tue, 22 Oct 2013 15:17:56 +0000 (15:17 +0000)]
Add an 'R' option to Dial which sends ringing until early media has been received.

(closes issue ASTERISK-10487)
Reported by: Gaspar Zoltan
Patches:
10487.patch uploaded by n8ideas (license 6075)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401411 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoRemove a noisy debug message from bridging code.
Mark Michelson [Mon, 21 Oct 2013 21:06:41 +0000 (21:06 +0000)]
Remove a noisy debug message from bridging code.

This particular debug message, during a stress test, was logged so
often that it appeared that there may be a memory leak in the logger
code. In actuality, there was no memory leak, but the logger thread
was having a hard time keeping up with the demands of the rest of the
system.

Since this debug message has no value at all, the best way to fix the
problem was to just remove the message.

(closes issue AST-1225)
reported by John Bigelow

Patches:
spammy_log.diff uploaded by Mark Michelson (License #5049)
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Merged revisions 401364 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401365 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoSegfault in LIBEDIT_INTERNAL after tgetstr(), when libncurses5-dev
Kevin Harwell [Mon, 21 Oct 2013 19:50:28 +0000 (19:50 +0000)]
Segfault in LIBEDIT_INTERNAL after tgetstr(), when libncurses5-dev
isn't installed

Include the appropriate declarations when not using termcap, but term+curses
and [n]curses do not exist.

(closes issue ASTERISK-22351)
Reported by: A. Iglesias
Patches:
    issueA22351_libedit_internal_without_ncurses_dev.patch uploaded by wdoekes (license 5674)
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Merged revisions 401326 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401327 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401328 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoFixing r401281; the model name is Channel, with a capital C
David M. Lee [Mon, 21 Oct 2013 18:59:51 +0000 (18:59 +0000)]
Fixing r401281; the model name is Channel, with a capital C
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Merged revisions 401315 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agoFixed malformed Access-Control-Allow-Methods header. Was causing Safari to barf on...
David M. Lee [Mon, 21 Oct 2013 18:59:22 +0000 (18:59 +0000)]
Fixed malformed Access-Control-Allow-Methods header. Was causing Safari to barf on POST and DELETE.
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Merged revisions 401106 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401316 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoFix IAX2 incoming call address lookups
Kinsey Moore [Sat, 19 Oct 2013 21:57:07 +0000 (21:57 +0000)]
Fix IAX2 incoming call address lookups

This fixes address lookup for incoming calls without a peer definition.
The address family was unset instead of being set to AST_AF_UNSPEC
which was causing lookup failures on "127.0.0.1". This is one of the
causes of the current failure of the app_page integration test.

Review: https://reviewboard.asterisk.org/r/2933/
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Merged revisions 401291 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401292 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoReturn a channel snapshot when originating using ARI, and subscribe the Stasis applic...
Joshua Colp [Sat, 19 Oct 2013 14:45:14 +0000 (14:45 +0000)]
Return a channel snapshot when originating using ARI, and subscribe the Stasis application to it.

This change allows a user of ARI to know what channel it has originated and also follow any
progress. If a Stasis application is provided it will be automatically subscribed to the
originated channel immediately.

(closes issue ASTERISK-22485)
Reported by: David Lee

Review: https://reviewboard.asterisk.org/r/2910/
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Merged revisions 401281 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401282 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_parking: Remove setting useless flag.
Richard Mudgett [Fri, 18 Oct 2013 22:52:35 +0000 (22:52 +0000)]
res_parking: Remove setting useless flag.
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Merged revisions 401271 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401272 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoThis is just a quick script for dumping swagger-ui into static-http,
David M. Lee [Fri, 18 Oct 2013 21:51:01 +0000 (21:51 +0000)]
This is just a quick script for dumping swagger-ui into static-http,
so that it can be served by the Asterisk web server.

I had to change the Makefile in order to recursively install content
from the static-http directory, hence the code review instead of just
putting it in.

Review: https://reviewboard.asterisk.org/r/2924/
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Merged revisions 401261 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401263 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoResolve some memory leaks due to incorrect for loop / ao2 ref usage.
Mark Michelson [Fri, 18 Oct 2013 18:44:21 +0000 (18:44 +0000)]
Resolve some memory leaks due to incorrect for loop / ao2 ref usage.

A common idiom in Asterisk is to due something like:

for (ao2_obj = list_beginning; ao2_obj = next_item; ao2_ref(ao2_obj, -1)) {
    ...do stuff...
}

This is nice because it automatically takes care of the object references
for you. However, there is a pitfall here. If a break statement is in the
for loop, then the current reference is not cleaned up. In some cases, this
is on purpose, but in others there is a leak. This commit fixes the leak
cases.
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Merged revisions 401248 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401249 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoAdd channel lock protection around translation path setup.
Richard Mudgett [Fri, 18 Oct 2013 16:59:09 +0000 (16:59 +0000)]
Add channel lock protection around translation path setup.

Most callers of ast_channel_make_compatible() happen before the channels
enter a two party bridge.  With the new bridging framework, two party
bridging technologies may also call ast_channel_make_compatible() when
there is more than one thread involved with the two channels.

* Added channel lock protection in set_format() and
ast_channel_make_compatible_helper() when dealing with the channel's
native formats while setting up a translation path.

* Fixed best_src_fmt and best_dst_fmt usage consistency in
ast_channel_make_compatible_helper().  The call to
ast_translator_best_choice() got them backwards.

* Updated some callers of ast_channel_make_compatible() and the function
documentation.  There is actually a difference between the two channels
passed in.

* Fixed the deadlock potential in res_fax.c dealing with
ast_channel_make_compatible().  The deadlock potential was already there
anyway because res_fax called ast_channel_make_compatible() with chan
locked.

(closes issue ASTERISK-22542)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2915/
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5 years agoTweak ast_bridge_depart() doxygen.
Richard Mudgett [Fri, 18 Oct 2013 16:20:54 +0000 (16:20 +0000)]
Tweak ast_bridge_depart() doxygen.
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5 years agoRemove the bit about requiring ast_bridge_depart() to be called before ast_bridge_des...
Mark Michelson [Fri, 18 Oct 2013 16:06:20 +0000 (16:06 +0000)]
Remove the bit about requiring ast_bridge_depart() to be called before ast_bridge_destroy().
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5 years agoClarify in ast_bridge_destroy() about how departable channels must be handled.
Mark Michelson [Fri, 18 Oct 2013 15:29:15 +0000 (15:29 +0000)]
Clarify in ast_bridge_destroy() about how departable channels must be handled.
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5 years agoRemove Port Restriction When Checking For NAT
Michael L. Young [Fri, 18 Oct 2013 15:14:36 +0000 (15:14 +0000)]
Remove Port Restriction When Checking For NAT

When trying to determine if a peer is behind NAT, we should not be using the
ports when comparing addresses.

This patch removes the port from being checked and just useds the addresses
now.

(closes issue ASTERISK-22729)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
    asterisk-remove-using-port-for-nat-check.diff
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2927/
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5 years agoProperly copy/remove the device state cache flag over a masquerade.
Walter Doekes [Fri, 18 Oct 2013 14:50:27 +0000 (14:50 +0000)]
Properly copy/remove the device state cache flag over a masquerade.

In r378303 the AST_FLAG_DISABLE_DEVSTATE_CACHE flag was added that tells
the devstate system to not cache states for non-real devices. However,
when optimizing away channels (ast_do_masquerade), that flag wasn't
copied.

In my case, using Local devices as queue members created a situation
where the endpoint was considered in use, but the state change of the
device being available again was ignored (not cached). The endpoint
channel was optimized into the (previously) Local channel, but kept
the do-not-cache flag. The end result being that the queue member
apparently stayed in use forever.

(closes issue ASTERISK-22718)
Reported by: Walter Doekes

Review: https://reviewboard.asterisk.org/r/2925/
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5 years agoFix Setting A chan_sip Dialog's SIP_NAT_FORCE_RPORT Flag
Michael L. Young [Thu, 17 Oct 2013 20:39:29 +0000 (20:39 +0000)]
Fix Setting A chan_sip Dialog's SIP_NAT_FORCE_RPORT Flag

A condition was added in a commit to fix ASTERISK-21374, that, if the
SIP_PAGE3_NAT_AUTO_RPORT flag was set, to then copy a peer's SIP_NAT_FORCE_RPORT
flag to the dialog.  This condition should not have been there since it assumed
that if Asterisk is in an environment where NAT is involved, that the auto_* nat
settings or force_rport setting would be on in the global settings.  If the nat
setting in the global setting is set to 'nat=no' and then turned on for peers
(which is not quite the recommended way, although it is allowed) this flag is
never copied to the dialog resulting in problems like, REGISTER replies going
to the wrong port.

This patch removes this conditional check and will now always use the peer's
flag which by this point in the code the checks on whether the peer is behind
NAT or not (if using auto_force_rport) have already been run.

(closes issue ASTERISK-22236)
Reported by: Filip Frank
Tested by: Michael L. Young
Patches:
    asterisk-2236-always-set-rport.diff uploaded
                                              by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2919/
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Merged revisions 401168 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agores_parking: Fix bug where reloading immediately wipes new parkpos extensions
Jonathan Rose [Thu, 17 Oct 2013 18:25:35 +0000 (18:25 +0000)]
res_parking: Fix bug where reloading immediately wipes new parkpos extensions

(closes issue ASTERISK-22631)
Reported by: Kevin Harwell
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5 years agoReduce log level of a non-pubsub error message
Kinsey Moore [Thu, 17 Oct 2013 15:41:22 +0000 (15:41 +0000)]
Reduce log level of a non-pubsub error message

Drop an error log message to debug level 1 since distributed device
state functions correctly when receiving this message and it spams the
logs.

(closes issue ASTERISK-22410)
Reported by: abelbeck
Patches:
    asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903)
    asterisk-11-res_xmpp-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903)
........

Merged revisions 401119 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401120 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401121 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agoARI: Fix crash when POST /playback/{id}/control does not have an operation parameter.
Richard Mudgett [Wed, 16 Oct 2013 21:22:25 +0000 (21:22 +0000)]
ARI: Fix crash when POST /playback/{id}/control does not have an operation parameter.

(closes issue ASTERISK-22680)
Reported by: John Bigelow
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5 years agoOops. Leftover /stasis reference
David M. Lee [Wed, 16 Oct 2013 17:01:28 +0000 (17:01 +0000)]
Oops. Leftover /stasis reference
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5 years agoClarify documentation for channel and bridge list
Kinsey Moore [Wed, 16 Oct 2013 14:02:06 +0000 (14:02 +0000)]
Clarify documentation for channel and bridge list

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

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5 years agoDon't check all realtime queues when doing "queue show some_queue".
Walter Doekes [Wed, 16 Oct 2013 12:19:47 +0000 (12:19 +0000)]
Don't check all realtime queues when doing "queue show some_queue".

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/
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Merged revisions 401049 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401076 from http://svn.asterisk.org/svn/asterisk/branches/11
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5 years agoUse POST / DELETE to toggle ARI bridge moh
Paul Belanger [Wed, 16 Oct 2013 00:12:36 +0000 (00:12 +0000)]
Use POST / DELETE to toggle ARI bridge moh

Review: https://reviewboard.asterisk.org/r/2911/
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5 years agotranslate.c: Some minor code tweaks.
Richard Mudgett [Tue, 15 Oct 2013 23:44:11 +0000 (23:44 +0000)]
translate.c: Some minor code tweaks.

* 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

5 years agobridge_native_dahdi: Return channel join failure if could not make the channels compa...
Richard Mudgett [Tue, 15 Oct 2013 20:26:13 +0000 (20:26 +0000)]
bridge_native_dahdi: Return channel join failure if could not make the channels compatible.
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5 years agochan_iax2: Fix channel left locked in off nominal code path.
Richard Mudgett [Tue, 15 Oct 2013 20:05:47 +0000 (20:05 +0000)]
chan_iax2: Fix channel left locked in off nominal code path.
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5 years agoEnsure bridge record error responses validate
Kinsey Moore [Tue, 15 Oct 2013 20:03:19 +0000 (20:03 +0000)]
Ensure bridge record error responses validate

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
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5 years agoUse POST / DELETE to toggle hold / moh for ARI channels
Paul Belanger [Tue, 15 Oct 2013 15:30:39 +0000 (15:30 +0000)]
Use POST / DELETE to toggle hold / moh for ARI channels

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/
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5 years agoPrevent chan_sip from sending duplicate BYEs.
Mark Michelson [Tue, 15 Oct 2013 15:26:06 +0000 (15:26 +0000)]
Prevent chan_sip from sending duplicate BYEs.

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
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Merged revisions 400971 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 400984 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agoMy doc correction in r400842 had a silly bug.
David M. Lee [Tue, 15 Oct 2013 13:44:45 +0000 (13:44 +0000)]
My doc correction in r400842 had a silly bug.

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)
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5 years agochan_dahdi: Add config support for hwgain settings.
Richard Mudgett [Mon, 14 Oct 2013 22:52:42 +0000 (22:52 +0000)]
chan_dahdi: Add config support for hwgain settings.

* 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

5 years agochan_dahdi: Reflect the set software gain in the CLI "dahdi show channel" output.
Richard Mudgett [Mon, 14 Oct 2013 22:06:01 +0000 (22:06 +0000)]
chan_dahdi: Reflect the set software gain in the CLI "dahdi show channel" output.

* 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
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5 years agochan_sip: Do not increment the SDP version between 183 and 200 responses.
Mark Michelson [Mon, 14 Oct 2013 22:03:22 +0000 (22:03 +0000)]
chan_sip: Do not increment the SDP version between 183 and 200 responses.

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)
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5 years agopjsip outbound registration: Log message says received a 408 when we didn't
Kevin Harwell [Mon, 14 Oct 2013 15:54:06 +0000 (15:54 +0000)]
pjsip outbound registration: Log message says received a 408 when we didn't

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

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5 years agoRemove duplicate module info block
Matthew Jordan [Mon, 14 Oct 2013 15:01:59 +0000 (15:01 +0000)]
Remove duplicate module info block

The module info block was repeated twice. Once is sufficient.
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5 years agoFix a race condition in res_pjsip_session with rapidly terminating the session.
Joshua Colp [Sun, 13 Oct 2013 15:42:20 +0000 (15:42 +0000)]
Fix a race condition in res_pjsip_session with rapidly terminating the session.

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

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5 years agoFix realm comparison for outbound auth
Kinsey Moore [Sat, 12 Oct 2013 16:53:06 +0000 (16:53 +0000)]
Fix realm comparison for outbound auth

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

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5 years agochannel.h: whitespace changes.
Richard Mudgett [Fri, 11 Oct 2013 17:05:01 +0000 (17:05 +0000)]
channel.h: whitespace changes.
........

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5 years agoMultiple revisions 400508,400842-400843,400848
David M. Lee [Fri, 11 Oct 2013 16:36:00 +0000 (16:36 +0000)]
Multiple revisions 400508,400842-400843,400848

........
  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)
........

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5 years agoFixed merge tracking for r400360, which was somehow lost
David M. Lee [Fri, 11 Oct 2013 16:33:14 +0000 (16:33 +0000)]
Fixed merge tracking for r400360, which was somehow lost

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

5 years agoSoftmix: Fix crash when switching from softmix to another bridge technology.
Richard Mudgett [Fri, 11 Oct 2013 16:28:50 +0000 (16:28 +0000)]
Softmix: Fix crash when switching from softmix to another bridge technology.

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
........

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5 years agoPerform validation of permanent contacts on AORs in res_pjsip.
Joshua Colp [Thu, 10 Oct 2013 18:21:55 +0000 (18:21 +0000)]
Perform validation of permanent contacts on AORs in res_pjsip.
........

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5 years agoFix an assertion in res_pjsip when specifying an invalid outbound proxy.
Joshua Colp [Thu, 10 Oct 2013 12:26:20 +0000 (12:26 +0000)]
Fix an assertion in res_pjsip when specifying an invalid outbound proxy.

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

5 years agoUse 'z' as the format specifier for size_t
Matthew Jordan [Wed, 9 Oct 2013 11:02:04 +0000 (11:02 +0000)]
Use 'z' as the format specifier for size_t

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.
........

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5 years agoAdd PJSIP_HEADER function for manipulation of SIP headers in the PJSIP stack
Matthew Jordan [Tue, 8 Oct 2013 22:59:32 +0000 (22:59 +0000)]
Add PJSIP_HEADER function for manipulation of SIP headers in the PJSIP stack

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

5 years agoAdd warning when compiling with iODBC support
Kinsey Moore [Tue, 8 Oct 2013 22:33:31 +0000 (22:33 +0000)]
Add warning when compiling with iODBC support

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

5 years agoapp_agent_pool: Fix AMI/CLI AgentLogoff soft preventing agents from logging back in.
Richard Mudgett [Tue, 8 Oct 2013 21:20:19 +0000 (21:20 +0000)]
app_agent_pool: Fix AMI/CLI AgentLogoff soft preventing agents from logging back in.

* 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

5 years agoSwitch from using pjsip_strerror to pj_strerror.
Mark Michelson [Tue, 8 Oct 2013 20:52:04 +0000 (20:52 +0000)]
Switch from using pjsip_strerror to pj_strerror.

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

5 years agoapp_confbridge: Can now set the language used for announcements to the conference.
Richard Mudgett [Tue, 8 Oct 2013 20:18:37 +0000 (20:18 +0000)]
app_confbridge: Can now set the language used for announcements to the conference.

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

5 years agoapp_confbridge: Fix duplicate default_user profile.
Richard Mudgett [Tue, 8 Oct 2013 19:18:05 +0000 (19:18 +0000)]
app_confbridge: Fix duplicate default_user profile.

* 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

5 years agoFix func_config list entry allocation
Kinsey Moore [Tue, 8 Oct 2013 18:19:59 +0000 (18:19 +0000)]
Fix func_config list entry allocation

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
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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

5 years agoFix STUN crash when using IPv6 any address
Kinsey Moore [Tue, 8 Oct 2013 15:46:16 +0000 (15:46 +0000)]
Fix STUN crash when using IPv6 any address

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
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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

5 years agoPush CLI qualify into the threadpool.
Mark Michelson [Tue, 8 Oct 2013 15:44:47 +0000 (15:44 +0000)]
Push CLI qualify into the threadpool.

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

5 years agoMake app_queue and res_agi independent of AMI being enabled.
Richard Mudgett [Tue, 8 Oct 2013 15:12:46 +0000 (15:12 +0000)]
Make app_queue and res_agi independent of AMI being enabled.

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

5 years agoMiscellaneous stand alone comment cleanups.
Richard Mudgett [Mon, 7 Oct 2013 15:43:22 +0000 (15:43 +0000)]
Miscellaneous stand alone comment cleanups.
........

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

5 years agoapp_queue: Fix Queuelog EXITWITHKEY only logging two of four fields
Michael L. Young [Sun, 6 Oct 2013 17:13:21 +0000 (17:13 +0000)]
app_queue: Fix Queuelog EXITWITHKEY only logging two of four fields

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/
........

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

5 years agochan_iax2: Fix compile error.
Richard Mudgett [Sat, 5 Oct 2013 00:59:17 +0000 (00:59 +0000)]
chan_iax2: Fix compile error.
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Merged revisions 400588 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agoAdd IPv6 Support To chan_iax2
Michael L. Young [Fri, 4 Oct 2013 21:41:58 +0000 (21:41 +0000)]
Add IPv6 Support To chan_iax2

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

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5 years agoAdded missing file from r400522
David M. Lee [Fri, 4 Oct 2013 19:32:29 +0000 (19:32 +0000)]
Added missing file from r400522
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5 years agochan_pjsip: Make logger togglable without loading/unloading
Jonathan Rose [Fri, 4 Oct 2013 19:11:38 +0000 (19:11 +0000)]
chan_pjsip: Make logger togglable without loading/unloading

This patch makes the res_pjsip_logger do a few things... First, it
will be built and installed by default now, so end users won't need
to enable it in menuselect. Second, while it is loaded, it no longer
will immediately issue log messages. Upon loading, it is in the
disabled state and must be turned on with the new CLI command. The
CLI command 'pjsip set logger <on/off/host> has been added and can be
used to do the following:
pjsip set logger on:
    Enables logger for all PJSIP traffic
pjsip set logger off:
    Disables logger for all PJSIP traffic
pjsip set logger host <host>:
    Enables logger for the specific host

Review: https://reviewboard.asterisk.org/r/2900/
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Merged revisions 400542 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400543 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agochan_pjsip: Add alembic scripts for generating db tables for PJSIP
Jonathan Rose [Fri, 4 Oct 2013 18:13:37 +0000 (18:13 +0000)]
chan_pjsip: Add alembic scripts for generating db tables for PJSIP

Also updates sample configurations for sorcery and extconfig to
demonstrate how to use databases created by that alembic script.

(closes issue ASTERISK-22133)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2892/
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