Commit Graph

24996 Commits

Author SHA1 Message Date
David M. Lee
2d0fb7f613 ari wiki docs: add notes about allowMultiple parameters.
This patch adds a note to any parameter that has 'allowMultiple' set in
the Swagger documentation.

(closes issue ASTERISK-22704)
........

Merged revisions 402367 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-01 16:31:57 +00:00
Joshua Colp
7678fd040e res_ari_channels: Add ring operation, dtmf operation, hangup reasons, and tweak early media.
The ring operation sends ringing to the specified channel it is invoked on.
The dtmf operation can be used to send DTMF digits to the specified channel
of a specific length with a wait time in between. Finally hangup reasons
allow you to specify why a channel is being hung up (busy, congestion).

Early media behavior has also been tweaked slightly. When playing media to a channel
it will no longer automatically answer. If it has not been answered a progress indication
is sent instead.

(closes issue ASTERISK-22701)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2916/
........

Merged revisions 402358 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-01 14:38:21 +00:00
Kinsey Moore
98dea21bc1 chan_sip: Fix RTCP port for SRFLX ICE candidates
This corrects one-way audio between Asterisk and Chrome/jssip as a
result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX
ICE candidates. This also exposes an ICE component enumeration to
extract further details from candidates.

(closes issue ASTERISK-21383)
Reported by: Shaun Clark
Review: https://reviewboard.asterisk.org/r/2967/
........

Merged revisions 402345 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 402348 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-01 12:40:40 +00:00
Joshua Colp
4053f36a71 res_ari_channels: Fix a deadlock when originating multiple channels close to eachother.
If a Stasis application is specified an implicit subscription is done on the originated
channel. This was previously done with the channel lock held which is dangerous as the
underlying code locks the container and iterates items. This change releases the lock
on the originated channel before subscribing occurs.

(closes issue ASTERISK-22768)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2979/
........

Merged revisions 402346 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-01 12:33:09 +00:00
Joshua Colp
d17a780333 res_stasis: Ensure the channel is always departed from the bridge when it leaves.
This change adds a command to the command queue to explicitly depart the channel
from the bridge when it is told it has left. If the channel has already been departed
or has entered a different bridge this command will become a no-op.

(closes issue ASTERISK-22703)
Reported by: John Bigelow

(closes issue ASTERISK-22634)
Reported by: Kevin Harwell

Review: https://reviewboard.asterisk.org/r/2965/
........

Merged revisions 402336 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402337 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-01 12:13:09 +00:00
Mark Michelson
dd221c74c5 Update the conversion script from sip.conf to pjsip.conf
(closes issue ASTERISK-22374)
Reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/2846
........

Merged revisions 402327 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-31 22:09:47 +00:00
Matthew Jordan
e9fc321053 core/loader: Don't call dlclose in a while loop
For awhile now, we've noticed continuous integration builds hanging on CentOS 6
64-bit build agents. After resolving a number of problems with symbols, strange
locks, and other shenanigans, the problem has persisted. In all cases, gdb
shows the Asterisk process stuck in loader.c on one of the infinite while loops
that calls dlclose repeatedly until success.

The documentation of dlclose states that it returns 0 on success; any other
value on error. It does not state that repeatedly calling it will eventually
clear those errors. Most likely, the repeated calls to dlclose was to force a
close by exhausting the references on the library; however, that will never
succeed if:
(a) There is some fundamental error at work in the loaded library that
    precludes unloading it
(b) Some other loaded module is referencing a symbol in the currently loaded
    module

This results in Asterisk sitting forever.

Since we have matching pairs of dlopen/dlclose, this patch opts to only call
dlclose once, and log out as an ERROR if dlclose fails to return success. If
nothing else, this might help to determine why on the CentOS 6 64-bit build agent
things are not closing successfully.

Review: https://reviewboard.asterisk.org/r/2970
........

Merged revisions 402287 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 402288 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 402289 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-31 16:06:14 +00:00
Matthew Jordan
981983bfde 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.
........

Merged revisions 402285 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-31 15:52:32 +00:00
David M. Lee
069da1e75a 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.
........

Merged revisions 402276 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402277 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-31 14:45:03 +00:00
Kevin Harwell
4746c068dc pjsip_messaging: Added debug for in dialog messaging
(issue ASTERISK-22777)
Reported by: Matt Jordan
........

Merged revisions 402265 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-30 17:54:26 +00:00
Rusty Newton
74dc7983d1 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)
........

Merged revisions 402224 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 402225 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 402226 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-29 23:43:58 +00:00
Matthew Jordan
076b29dd5b 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/
........

Merged revisions 402150 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 402151 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 402154 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-29 12:57:35 +00:00
Kinsey Moore
aa7f9e55f2 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/
........

Merged revisions 402152 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402153 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-29 12:51:57 +00:00
Kinsey Moore
6c417b0475 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)
........

Merged revisions 402148 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402149 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-29 12:30:21 +00:00
Joshua Colp
fd98037fe2 res_ari_playback: Add missing 404 error response for GET and DELETE.
(closes issue ASTERISK-22722)
Reported by: Richard Mudgett
........

Merged revisions 402139 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-29 11:15:59 +00:00
David M. Lee
ef57947f6d Ignore full docs
........

Merged revisions 402127 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-28 22:10:16 +00:00
David M. Lee
0d9e4fa817 Put back several merge revisions that were lost in r402054
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402129 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-28 22:09:49 +00:00
David M. Lee
13c645ef02 Put back several merge revisions that were lost in r401962
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402128 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-28 22:05:37 +00:00
Michael L. Young
fead41381b 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.
........

Merged revisions 402115 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-28 15:08:00 +00:00
Michael L. Young
230141d677 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
........

Merged revisions 402111 from http://svn.asterisk.org/svn/asterisk/branches/11
........

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
2013-10-28 14:59:16 +00:00
Matthew Jordan
26182f4b71 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.
........

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
2013-10-27 23:22:51 +00:00
Matthew Jordan
3713fa5c9f 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/
........

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
2013-10-27 20:04:17 +00:00
Matthew Jordan
2e24dfe4d1 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.
........

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
2013-10-27 02:39:34 +00:00
Joshua Colp
26914adc00 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
........

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
2013-10-26 12:56:08 +00:00
Richard Mudgett
8e4084b586 res_stasis.c: Made use the ao2_container callback templates.
* Made res_stasis.c use the OBJ_SEARCH_XXX defines.
........

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
2013-10-26 00:36:31 +00:00
Scott Griepentrog
39a233d32b 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/
........

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
........

Merged revisions 402042 from http://svn.asterisk.org/svn/asterisk/branches/11
........

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
2013-10-26 00:27:02 +00:00
Richard Mudgett
78790f0d58 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.
........

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
2013-10-25 23:58:32 +00:00
Richard Mudgett
7eea4ab872 You'd think that new files would be free of whitespace issues. But you would be wrong.
........

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
2013-10-25 22:03:04 +00:00
Jonathan Rose
5c696bde67 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/
........

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
2013-10-25 22:01:43 +00:00
Jonathan Rose
d8a760307e 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/
........

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
2013-10-25 21:28:32 +00:00
Scott Griepentrog
7b42a6828a 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/
........

Merged revisions 401959 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 401960 from http://svn.asterisk.org/svn/asterisk/branches/11
........

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
2013-10-25 20:51:13 +00:00
Jonathan Rose
bb1568caa1 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/
........

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
2013-10-25 17:41:38 +00:00
Jonathan Rose
e920fc0669 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)
........

Merged revisions 401914 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 401935 from http://svn.asterisk.org/svn/asterisk/branches/11
........

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
2013-10-25 17:32:17 +00:00
Jonathan Rose
b7763842a7 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
........

Merged revisions 401895 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 401896 from http://svn.asterisk.org/svn/asterisk/branches/11
........

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
2013-10-25 16:59:33 +00:00
Kevin Harwell
9ba7742431 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)
........

Merged revisions 401884 from http://svn.asterisk.org/svn/asterisk/branches/11
........

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
2013-10-25 16:09:05 +00:00
David M. Lee
ce7cddc362 Blocked revisions 401391
........
Blocked revisions 401379

........
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.
........

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
2013-10-25 13:50:57 +00:00
David M. Lee
8f43c9716c 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/
........

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
2013-10-25 13:49:20 +00:00
Kevin Harwell
b7bb1de4d2 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
........

Merged revisions 401833 from http://svn.asterisk.org/svn/asterisk/branches/11
........

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
2013-10-24 21:06:14 +00:00
David M. Lee
32b4e79434 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/
........

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
2013-10-24 20:48:17 +00:00
Jonathan Rose
6fb07febbc 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)
........

Merged revisions 401829 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 401830 from http://svn.asterisk.org/svn/asterisk/branches/11
........

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
2013-10-24 20:34:53 +00:00
Jonathan Rose
1d0a6d2b2c 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)
........

Merged revisions 401790 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 401791 from http://svn.asterisk.org/svn/asterisk/branches/11
........

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
2013-10-24 19:57:04 +00:00
Jonathan Rose
d22fd3e3f6 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
........

Merged revisions 401786 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 401787 from http://svn.asterisk.org/svn/asterisk/branches/11
........

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
2013-10-24 19:42:21 +00:00
Jonathan Rose
95d8977e22 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)
........

Merged revisions 401781 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 401783 from http://svn.asterisk.org/svn/asterisk/branches/11
........

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
2013-10-24 19:31:23 +00:00
Jonathan Rose
713ac0872b 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
........

Merged revisions 401744 from http://svn.asterisk.org/svn/asterisk/branches/11
........

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
2013-10-24 18:46:56 +00:00
Jonathan Rose
4ca0f222e8 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
........

Merged revisions 401705 from http://svn.asterisk.org/svn/asterisk/branches/11
........

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
2013-10-24 17:00:27 +00:00
Jonathan Rose
beb5cdbef5 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)
........

Merged revisions 401660 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 401661 from http://svn.asterisk.org/svn/asterisk/branches/11
........

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
2013-10-23 20:10:30 +00:00
Jonathan Rose
d7bac6cf4b 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/
........

Merged revisions 401619 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 401620 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 401621 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-23 17:56:44 +00:00
Richard Mudgett
2966ca288c 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
........

Merged revisions 401577 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 401579 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 401581 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-23 16:52:11 +00:00
John Bigelow
3975617f87 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/
........

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
2013-10-23 16:48:39 +00:00
Kinsey Moore
dfd3f4ef46 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)
........

Merged revisions 401537 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 401538 from http://svn.asterisk.org/svn/asterisk/branches/11
........

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
2013-10-23 15:23:58 +00:00