Commit Graph

24790 Commits

Author SHA1 Message Date
Matthew Jordan
f8cd9b61f6 medix_index: Display errors when library calls fail
Based on feedback from ipengineer in #asterisk, when the media indexer
cannot access a sound file on the system (or otherwise fails) Asterisk
displays a "Cannot frob file" error but fails to tell you why. This is
especially problematic as the media_indexer failing will rpevent Asterisk
from starting, as it is in the core.

We now display the errno error messages so folks can figure out what they've
done wrong.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402285 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-31 15:51:36 +00:00
David M. Lee
a181e534e0 stasis: add functions embarrassingly missing from r400522
I neglected to implement two of the endpoint subscription functions when
I did the work. Normally, you'll only hit that when you unsubscribe from
a specific endpoint.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-31 14:43:44 +00:00
Kevin Harwell
8a08f73fe0 pjsip_messaging: Added debug for in dialog messaging
(issue ASTERISK-22777)
Reported by: Matt Jordan


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402265 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-30 17:52:55 +00:00
Rusty Newton
cd7f9cec0c Updates for 1.4.25 core sounds and 1.4.14 extra sounds, plus new en_GB language set
The new sound packages relate to issues: ASTERISK-22544, ASTERISK-22411, ASTERISK-21413, ASTERISK-20782
Modified sounds/Makefile for the new sound versions and to account for the new en_GB language set.

(issue ASTERISK-22659)
(closes issue ASTERISK-22659)
(closes issue ASTERISK-22411)
(closes issue ASTERISK-22544)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-29 23:43:18 +00:00
Matthew Jordan
837be45e9f Remove some spammy debug messages; improve clarity of others
Debug messages aren't free. Even when the debug level is sufficiently low such
that the messages are never evaluated, there is a cost to having to parse
Asterisk logs that contain debug messages that (a) fail to convey sufficient
information or (b) occur so frequently as to be next to meaningless. Based on
having to stare at lots of DEBUG messages, this patch makes the following
changes:

* channel.c: When copying variables from a parent channel to a child channel,
  specify the channels involved. Do not log anything for a variable that is not
  inherited; the fact that it doesn't have an _ or __ already signifies that it
  won't be inherited.
* pbx.c: Specify what function evaluation has occurred that created the result.
* translate.c: Bump up the translator path messages to 10. I've never once had
  to use these debug messages, and for each format that is registered (on
  startup) and unregistered (on shutdown) the entire f^2 matrix is logged out.
  For short tests in the Asterisk Test Suite, this should make finding the
  actual test much easier.
* xmldoc.c: The debug message that 'blah' is not found in the tree is expected.
  Often, description elements - which are not required - are not provided.
  This debug message adds no additional value, as it is not indicative of an
  error or helpful in debugging which element did not contain a 'blah' element
  as a child. If an element is supposed to contain a child element, then that
  XML tree should have failed validation in the first place.

Review: https://reviewboard.asterisk.org/r/2966/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-29 12:53:17 +00:00
Kinsey Moore
e81d1ab8cd ARI: Remove channels/{channelId}/dial
This removes the /ari/channels/{channelId}/dial URI since it is
redundant, overly complex, is likely to become more externally complex
over time, and is too high-level compared with other ARI operations.
See the following for further information:
http://lists.digium.com/pipermail/asterisk-app-dev/2013-October/000002.html

(closes issue ASTERISK-22784)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2968/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-29 12:51:15 +00:00
Kinsey Moore
67006871d6 bridge_native_rtp: Ensure bridge is torn down
When a bridge transitions away from one tech to another, the tech going
away is provided a dummy bridge with no channels in it to tear down.
Currently this means that the teardown code exits prematurely and does
not tear anything down. This change tears down RTP bridging for the
channel provided in the leave bridge tech callback.

This also reverts the majority of r400403 since it is now redundant.

(closes issue ASTERISK-22628)
(closes issue ASTERISK-22676)
Reported by: John Bigelow
Reported by: Kevin Harwell
Tested by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2905/
Patches:
    native_rtp_fix.diff uploaded by Kinsey Moore (License 6273)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-29 12:26:49 +00:00
Joshua Colp
8de298e17b res_ari_playback: Add missing 404 error response for GET and DELETE.
(closes issue ASTERISK-22722)
Reported by: Richard Mudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-29 11:15:16 +00:00
David M. Lee
c5754c698c Ignore full docs
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-28 21:30:37 +00:00
Michael L. Young
665e1bfb24 Fix UPGRADE.txt Due To Merging From Branch 11
When merging in the patch for ASTERISK-22728, the UPGRADE.txt file was changed
incorrectly.  That change should have gone into ASTERISK-11.txt.

This commit is to fix that.

Also, another comment in the UPGRADE-11.txt was missing and this commit adds
that as well.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-28 15:05:48 +00:00
Michael L. Young
7e61a3028e 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-28 14:51:55 +00:00
Matthew Jordan
4faa10f44a 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.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-27 23:22:19 +00:00
Matthew Jordan
5740a2bda6 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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-27 19:40:43 +00:00
Joshua Colp
23be89dfff 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


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-26 12:55:11 +00:00
Richard Mudgett
ca09cb657c res_stasis.c: Made use the ao2_container callback templates.
* Made res_stasis.c use the OBJ_SEARCH_XXX defines.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-26 00:34:25 +00:00
Richard Mudgett
4ae14324aa 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.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-25 23:52:13 +00:00
Scott Griepentrog
330a2b4177 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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Test shows rtpmap:119 being copied per this change, but is not in sip invite
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-25 23:48:06 +00:00
Richard Mudgett
d1e6a11deb You'd think that new files would be free of whitespace issues. But you would be wrong.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-25 22:02:31 +00:00
Jonathan Rose
0bdbdbf3fc 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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@402001 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-25 21:53:31 +00:00
Jonathan Rose
f503fc5673 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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-25 21:20:42 +00:00
Scott Griepentrog
1de75f4fae 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401961 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-25 20:47:55 +00:00
Jonathan Rose
155ba34659 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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-25 17:34:30 +00:00
Jonathan Rose
a5ca8e9347 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-25 17:31:38 +00:00
Jonathan Rose
b5cb5a0aa4 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-25 16:57:58 +00:00
Kevin Harwell
9afa2ff076 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401885 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-25 16:07:38 +00:00
David M. Lee
e8b74d4d57 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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401872 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-25 13:48:40 +00:00
Kevin Harwell
27334b9093 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-24 20:56:50 +00:00
Jonathan Rose
2e3908a554 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-24 20:34:23 +00:00
Jonathan Rose
23a4856d63 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-24 19:56:06 +00:00
Jonathan Rose
09bea85f3d 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-24 19:41:49 +00:00
Jonathan Rose
8f5bad357b 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401784 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-24 19:30:50 +00:00
Jonathan Rose
b2ea7c51ed 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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401745 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-24 18:45:52 +00:00
Jonathan Rose
b5cc4ae955 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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-24 16:58:30 +00:00
David M. Lee
5f31d0b182 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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-24 03:12:06 +00:00
Jonathan Rose
77136643d4 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-23 20:02:21 +00:00
Jonathan Rose
ac7cd94477 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/branches/12@401621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-23 17:45:36 +00:00
Richard Mudgett
f855ba2a76 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/branches/12@401581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-23 16:49:51 +00:00
John Bigelow
6c89018948 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/



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-23 16:37:34 +00:00
Kinsey Moore
b0b7db475c 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-23 15:23:30 +00:00
Joshua Colp
de0ec33701 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-23 11:14:49 +00:00
Michael L. Young
ecb2759060 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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-23 02:31:48 +00:00
Matthew Jordan
d8bda6cf59 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22 22:50:23 +00:00
Richard Mudgett
5d783eff54 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22 19:03:02 +00:00
Richard Mudgett
aa94b9c148 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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22 17:05:14 +00:00
Richard Mudgett
420b5e27db 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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401422 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22 16:32:23 +00:00
Richard Mudgett
51adbac546 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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401420 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22 16:26:20 +00:00
Richard Mudgett
a3a6f6bc38 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401391 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22 00:20:14 +00:00
Mark Michelson
61a061aa3c 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)



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-21 21:05:37 +00:00
Kevin Harwell
2a16b55404 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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2013-10-21 19:48:28 +00:00
David M. Lee
222e34644f Fixing r401281; the model name is Channel, with a capital C
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401315 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-21 18:58:37 +00:00