[asterisk-commits] file: branch file/sorceryx3 r386185 - in /team/file/sorceryx3: ./ apps/ build...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Sat Apr 20 11:12:05 CDT 2013
Author: file
Date: Sat Apr 20 11:11:43 2013
New Revision: 386185
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=386185
Log:
Multiple revisions 383611,383632-383633,383669,383726,383728,383747,383753-383754,383799,383837-383838,383841,383879,383925,383948,383975,383980,384019,384050,384120,384164,384201,384219,384261,384302,384327,384389-384390,384412-384413,384416,384452,384488,384514,384518,384546,384616,384642,384696,384711,384760,384828,384857,384879,384910,384942,384989,385049,385088,385116,385142,385174,385202,385236,385277-385278,385314,385357,385406,385431,385474,385522,385548,385573,385585,385595,385635,385638,385718,385734,385742-385743,385782,385835,385860,385862,385886,385939,386019-386020,386054,386160
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r383611 | dlee | 2013-03-22 16:26:37 -0300 (Fri, 22 Mar 2013) | 6 lines
Corrected some module issues introduced by r383579.
When I moved res_json.c to json.c, I left the MODULE_INFO stuff in there,
which was interesting if you ran module show. I also forgot to call what
was in module_load() from asterisk main().
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r383632 | elguero | 2013-03-22 17:43:24 -0300 (Fri, 22 Mar 2013) | 21 lines
Fix StopMixMonitor Hanging Up When Unable To Stop MixMonitor On A Channel
A regression was accidentally introduced when allowing an optional ID to be used
when calling StopMixMonitor. When we are unable to stop MixMonitor on a
channel, -1 is being returned which triggers the hangup of the channel.
This patch restores the prior behavior by returning 0 whether we were successful
or not. It also allows the call from the manager to use the return code when
the action fails.
(closes issue ASTERISK-21294)
Reported by: daroz
Tested by: daroz
Patches:
asterisk-21294-stop_mixmonitor_hangingup.diff Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2404/
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Merged revisions 383631 from http://svn.asterisk.org/svn/asterisk/branches/11
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r383633 | dlee | 2013-03-22 17:51:33 -0300 (Fri, 22 Mar 2013) | 5 lines
Fixed another issue from r383579.
Core modules don't honor <depend> flags in MODULEINFO, which broke jansson
if specified --with-jansson to configure.
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r383669 | seanbright | 2013-03-25 09:38:15 -0300 (Mon, 25 Mar 2013) | 8 lines
Properly delimit post data in res_config_curl.
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Merged revisions 383667 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 383668 from http://svn.asterisk.org/svn/asterisk/branches/11
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r383726 | dlee | 2013-03-25 13:19:55 -0300 (Mon, 25 Mar 2013) | 28 lines
Move NewCallerid, HangupRequest and SoftHangupRequest to Stasis
HangupRequest and SoftHangupRequest are now ast_channel_blob Stasis
messages, with the cause code as an optional field in the blob.
NewCallerid now simply watches for changes in the callerid information
in channel snapshots, and creates the AMI event appropriately.
Since the original NewCallerid event honored the channelvars setting
in manager.conf, the channel variables configured there had to become
a part of the channel snapshot. These are now a part of every snapshot
based event, making the configuration description "every time a
channel-oriented event is emitted" less of a lie.
There a a few other changes wrapped up in here as well.
* When ast_channel_topic() is given NULL for a channel, it returns
the ast_channel_topic_all() topic instead of NULL. This can clean
up a lot of NULL checking we're doing currently.
* The fields Cause and Cause-txt were removed from the base channel
information and put only on the Hangup events, since those fields
are meaningless outside of a Hangup event.
* Removed the pipe-delimiter processing of the channelvars field,
since that's been deprecated forever.
(closes issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2405/
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r383728 | dlee | 2013-03-25 14:12:03 -0300 (Mon, 25 Mar 2013) | 1 line
install_prereq: Adding jansson-devel to RH packages
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r383747 | dlee | 2013-03-25 16:28:04 -0300 (Mon, 25 Mar 2013) | 1 line
install_prereq: removed some out-of-date comments
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r383753 | kmoore | 2013-03-25 17:07:00 -0300 (Mon, 25 Mar 2013) | 2 lines
Fix missing ' ' around '='
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r383754 | kmoore | 2013-03-25 17:15:09 -0300 (Mon, 25 Mar 2013) | 2 lines
Fix typo
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r383799 | rmudgett | 2013-03-25 20:25:32 -0300 (Mon, 25 Mar 2013) | 15 lines
Set the CALLERID(dnid-num-plan) for incoming ISDN calls.
The CALLEDTON channel variable is set for incoming ISDN calls to the lower
7 bits of the Q.931 type-of-number/numbering-plan octet. The
CALLERID(dnid-num-plan) should have the same value.
(closes issue ASTERISK-21248)
Reported by: rmudgett
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Merged revisions 383796 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 383798 from http://svn.asterisk.org/svn/asterisk/branches/11
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r383837 | russell | 2013-03-25 22:38:56 -0300 (Mon, 25 Mar 2013) | 19 lines
Fix multi-station answer race condition.
When an SLA trunk is ringing (inbound call on the trunk) Asterisk will
make outbound calls to the stations that have that trunk. If more than
one station answers the call at the same time, all channels other than
the first one to answer are left in a bad state. The channel gets
leaked, is not connected to anything, and there's no way to get rid of
it.
We now properly clean up these losing channels by hanging up on them.
Since they lost the race, as we process their answer, there is no
ringing trunk for them to answer.
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Merged revisions 383835 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 383836 from http://svn.asterisk.org/svn/asterisk/branches/11
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r383838 | russell | 2013-03-25 22:46:39 -0300 (Mon, 25 Mar 2013) | 7 lines
Suppress compiler warning.
This code caused a compiler warning when --enable-dev-mode was not used.
The warning was that this variable was set but not used. That was indeed
the case as the only place this is used is as an argument to SKINNY_DEBUG
which is compiled out when not in dev mode.
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r383841 | mjordan | 2013-03-25 22:58:45 -0300 (Mon, 25 Mar 2013) | 22 lines
Resolve deadlock between pending CDR and batch CDR locks
r375757 attempted to resolve a race condition between multiple submissions of
CDRs while in batch mode from attempting to destroy the scheduled batch
submission by extending the batch CDR lock. Unfortunately, this causes a
deadlock between the pending CDR lock and the batch CDR lock. This patch
resolves the intent of r375757 by simply providing a new lock that protects
the scheduling of the batches. The original batch CDR lock is kept to protect
manipulation of the batch CDR settings, but has been placed such that it
is not held when the pending lock is held.
Thanks to Chase Venters for providing lock analysis on the issue.
(issue ASTERISK-21162)
Reported by: Chase Venters
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Merged revisions 383839 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 383840 from http://svn.asterisk.org/svn/asterisk/branches/11
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r383879 | mjordan | 2013-03-25 23:30:10 -0300 (Mon, 25 Mar 2013) | 41 lines
Resolve deadlock between SIP registration and channel based functions
In r373424, several reentrancy problems in chan_sip were addressed. As a
result, the SIP channel driver is now properly locking the channel driver
private information in certain operations that it wasn't previously. This
exposed two latent problems either in register_verify or by functions called
by register_verify. This includes:
* Holding the private lock while calling sip_send_mwi_to_peer. This can create
a new sip_pvt via sip_alloc, which will obtain the channel container lock.
This is a locking inversion, as any channel related lock must be obtained
prior to obtaining the SIP channel technology private lock.
Note that this issue was already fixed in Asterisk 11.
* Holding the private lock while calling sip_poke_peer. In the same vein as
sip_send_mwi_to_peer, sip_poke_peer can create a new SIP private, causing
the same locking inversion.
Note that this locking inversion typically occured when CLI commands were run
while a SIP REGISTER request was being processed, as many CLI commands (such
as 'sip show channels', 'core show channels', etc.) have to obtain the channel
container lock.
(issue ASTERISK-21068)
Reported by: Nicolas Bouliane
(issue ASTERISK-20550)
Reported by: David Brillert
(issue ASTERISK-21314)
Reported by: Badalian Vyacheslav
(issue ASTERISK-21296)
Reported by: Gabriel Birke
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Merged revisions 383863 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 383878 from http://svn.asterisk.org/svn/asterisk/branches/11
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r383925 | file | 2013-03-26 20:34:43 -0300 (Tue, 26 Mar 2013) | 2 lines
Remove the noop handler from sorcery so it does not produce an empty value.
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r383948 | wedhorn | 2013-03-27 04:24:37 -0300 (Wed, 27 Mar 2013) | 12 lines
Fix skinny encall button to not blind xfer.
The softbutton endcall should not turn a transfer into a blind transfer but
hangup the exten being called and leave the original call on hold. This does
that.
(closes issue ASTERISK-21321)
Reported by: wedhorn
Tested by: snuffy, myself
Patches:
skinny-xferendcall01.diff uploaded by wedhorn (license 5019)
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r383975 | mjordan | 2013-03-27 11:28:36 -0300 (Wed, 27 Mar 2013) | 16 lines
AST-2013-001: Prevent buffer overflow through H.264 format negotiation
The format attribute resource for H.264 video performs an unsafe read against a
media attribute when parsing the SDP. The value passed in with the format
attribute is not checked for its length when parsed into a fixed length buffer.
This patch resolves the vulnerability by only reading as many characters from
the SDP value as will fit into the buffer.
(closes issue ASTERISK-20901)
Reported by: Ulf Harnhammar
patches:
h264_overflow_security_patch.diff uploaded by jrose (License 6182)
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Merged revisions 383973 from http://svn.asterisk.org/svn/asterisk/branches/11
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r383980 | mjordan | 2013-03-27 11:39:11 -0300 (Wed, 27 Mar 2013) | 24 lines
AST-2013-002: Prevent denial of service in HTTP server
AST-2012-014, fixed in January of this year, contained a fix for Asterisk's
HTTP server for a remotely-triggered crash. While the fix put in place fixed
the possibility for the crash to be triggered, a denial of service vector still
exists with that solution if an attacker sends one or more HTTP POST requests
with very large Content-Length values. This patch resolves this by capping
the Content-Length at 1024 bytes. Any attempt to send an HTTP POST with
Content-Length greater than this cap will not result in any memory allocation.
The POST will be responded to with an HTTP 413 "Request Entity Too Large"
response.
This issue was reported by Christoph Hebeisen of TELUS Security Labs
(closes issue ASTERISK-20967)
Reported by: Christoph Hebeisen
patches:
AST-2013-002-1.8.diff uploaded by mmichelson (License 5049)
AST-2013-002-10.diff uploaded by mmichelson (License 5049)
AST-2013-002-11.diff uploaded by mmichelson (License 5049)
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Merged revisions 383978 from http://svn.asterisk.org/svn/asterisk/branches/11
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r384019 | mjordan | 2013-03-27 12:27:31 -0300 (Wed, 27 Mar 2013) | 35 lines
AST-2013-003: Prevent username disclosure in SIP channel driver
When authenticating a SIP request with alwaysauthreject enabled, allowguest
disabled, and autocreatepeer disabled, Asterisk discloses whether a user
exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways. The
information is disclosed when:
* A "407 Proxy Authentication Required" response is sent instead of a
"401 Unauthorized" response
* The presence or absence of additional tags occurs at the end of "403
Forbidden" (such as "(Bad Auth)")
* A "401 Unauthorized" response is sent instead of "403 Forbidden" response
after a retransmission
* Retransmission are sent when a matching peer did not exist, but not when a
matching peer did exist.
This patch resolves these various vectors by ensuring that the responses sent
in all scenarios is the same, regardless of the presence of a matching peer.
This issue was reported by Walter Doekes, OSSO B.V. A substantial portion of
the testing and the solution to this problem was done by Walter as well - a
huge thanks to his tireless efforts in finding all the ways in which this
setting didn't work, providing automated tests, and working with Kinsey on
getting this fixed.
(closes issue ASTERISK-21013)
Reported by: wdoekes
Tested by: wdoekes, kmoore
patches:
AST-2013-003-1.8 uploaded by kmoore, wdoekes (License 6273, 5674)
AST-2013-003-10 uploaded by kmoore, wdoekes (License 6273, 5674)
AST-2013-003-11 uploaded by kmoore, wdoekes (License 6273, 5674)
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Merged revisions 384003 from http://svn.asterisk.org/svn/asterisk/branches/11
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r384050 | kmoore | 2013-03-27 14:07:44 -0300 (Wed, 27 Mar 2013) | 22 lines
Fix white noise on SRTP decryption
When res_rtp_asterisk.c was altered to avoid attempting to apply
unprotect algorithms to non-audio RTP packets, the test used was
incorrect. This caused the audio packets to not be decrypted and
resulted in loud white noise on the other endpoint (or both endpoints
depending on the call legs involved). The test now properly checks the
version field in the RTP header to ensure that RTP and RTCP are
decrypted while other types of packets are not.
(closes issue ASTERISK-21323)
Reported by: andrea
Tested by: Kinsey Moore, andrea, John Bigelow
Patches:
whitenoise_fix.diff uploaded by Kinsey Moore
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Merged revisions 384048 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r384120 | mjordan | 2013-03-27 15:52:16 -0300 (Wed, 27 Mar 2013) | 20 lines
Fix a file descriptor leak in off nominal path
While looking at the security vulnerability in ASTERISK-20967, Walter noticed
a file descriptor leak and some other issues in off nominal code paths. This
patch corrects them.
Note that this patch is not related to the vulnerability in ASTERISK-20967,
but the patch was placed on that issue.
(closes issue ASTERISK-20967)
Reported by: wdoekes
patches:
issueA20967_file_leak_and_unused_wkspace.patch uploaded by wdoekes (License 5674)
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Merged revisions 384118 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 384119 from http://svn.asterisk.org/svn/asterisk/branches/11
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r384164 | kmoore | 2013-03-27 16:52:19 -0300 (Wed, 27 Mar 2013) | 8 lines
Address uninitialized conditional that valgrind found
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Merged revisions 384162 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r384201 | dlee | 2013-03-27 18:52:43 -0300 (Wed, 27 Mar 2013) | 1 line
Added a doxygen group for Stasis messages and topics
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r384219 | kmoore | 2013-03-27 19:42:06 -0300 (Wed, 27 Mar 2013) | 2 lines
Convert MWI state message type to the new stasis naming convention
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r384261 | kmoore | 2013-03-28 12:45:18 -0300 (Thu, 28 Mar 2013) | 2 lines
Break the world. Stasis message type accessors should now all be named correctly.
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r384302 | rmudgett | 2013-03-28 20:59:20 -0300 (Thu, 28 Mar 2013) | 16 lines
Add uuid wrapper API call ast_uuid_generate_str().
* Updated test_uuid.c to test the new API call.
* Made system use the new API call to eliminate "10's of lines" where
used.
* Fixed untested ast_strdup() return in stasis_subscribe() by eliminating
the need for it. struct stasis_subscription now contains the uniqueid[]
string.
* Fixed some issues in exchangecal_write_event():
Create uid with enough space for a UUID string to avoid a realloc.
Fix off by one error if the calendar event provided a UUID string.
There is no need to check for NULL before calling ast_free().
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r384327 | jrose | 2013-03-29 13:37:23 -0300 (Fri, 29 Mar 2013) | 19 lines
app_voicemail: Add blank argument to externnotify if no context argument
At least one call to run_externnotify provides a NULL context parameter and
because the snprintf statement doesn't account for a NULL context parameter,
it simply writes '(null)' to the arguments string instead. This patch makes
it write two quotes back to back for that argument instead in the event of
a NULL context.
(closes issue ASTERISK-18207)
Reported by: Barry L. Kline
Patches:
modified from patch-20130306 uploaded by Karsten Wemheuer (License 5930)
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Merged revisions 384325 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r384389 | mjordan | 2013-03-30 02:06:54 -0300 (Sat, 30 Mar 2013) | 8 lines
Convert TestEvent AMI events over to Stasis Core
This patch migrates the TestEvent AMI events to first be dispatched over the
Stasis-Core message bus. This helps to preserve the ordering of the events
with other events in the AMI system, such as the various channel related
events.
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r384390 | mjordan | 2013-03-30 02:15:42 -0300 (Sat, 30 Mar 2013) | 2 lines
Properly format an intmax_t value
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r384412 | dlee | 2013-04-01 10:34:51 -0300 (Mon, 01 Apr 2013) | 19 lines
Fix parallel make problems.
Occasionally, make -j would fail due to missing includes, or other
unusual errors.
This was due to the 'cleantest' target, which was designed to force a
make clean when some change in the code would cause the typical
depedency checking to fail. Several targets in the main Makefile did
not depend upon cleantest, hence would run in parallel to it. By
adding the dependency, make -j runs happily now.
Review: https://reviewboard.asterisk.org/r/2418/
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Merged revisions 384410 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 384411 from http://svn.asterisk.org/svn/asterisk/branches/11
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r384413 | dlee | 2013-04-01 10:37:51 -0300 (Mon, 01 Apr 2013) | 22 lines
stasis: Fixed message ordering issues when forwarding
This patch fixes an issue of message ordering that occurs when
multiple topics are forwarded to an aggregator topic (such as
ast_channel_topic_all()).
It is (very reasonably) expected that the rules governing message
dispatch order still apply, so long as the messages start from the
same thread, and are received by the same subscription. Because the
existing code had an additional layer of dispatching via the Stasis
thread pool for forwards, those promises couldn't be kept.
Forwarding subscriptions no longer have their own mailbox, and now
dispatch directly from the forwarding topic's stasis_publish()
call. This means that the topic's lock is held for the duration of not
only a message's dispatch, but the dispatch of all the forwards. This
shouldn't be a problem right now, but if an aggregator topic had many
subscribers, it could become a problem. But I figure we can write more
clever code when the time comes, if necessary.
Review: https://reviewboard.asterisk.org/r/2419/
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r384416 | file | 2013-04-01 11:10:46 -0300 (Mon, 01 Apr 2013) | 5 lines
Remove silly use of strncmp.
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Merged revisions 384414 from http://svn.asterisk.org/svn/asterisk/branches/11
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r384452 | mjordan | 2013-04-01 11:44:30 -0300 (Mon, 01 Apr 2013) | 6 lines
Make appropriate items parse using '|' instead of ','
This patch fixes a bug introduced in r76703, wherein Asterisk could only parse
arguments in the so-called 'recommended' way, e.g., NoOp(foo,bar). The proper
syntax of NoOp,foo|bar is now parsed correctly.
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r384488 | dlee | 2013-04-01 17:10:47 -0300 (Mon, 01 Apr 2013) | 28 lines
install_prereq: Build jansson from source, when necessary
When r383579 was committed, it made Jansson a required dependency.
While libjansson-dev and jansson-devel are available on recent
distros, some older (but still supported) distros don't have
it. There's a pull request[1] to get it into repoforge, but that still
doesn't help everyone. (And helps no one until the pull request is
merged and packages are built).
This patch adds Jansson install from source to the install_unpackaged()
function. There are a few gotcha's, which makes this change not
completely trivial.
* Since Jansson may be installed by a package, don't install from
source if a package installation can be found
* libresample may also be installed via package, so I added a
similar check to that.
* Since Jansson installs into /usr/local, this patch also adds
/usr/local/lib to /etc/ld.so.conf.d so that the library can be
found.
* The alternative was to install into /usr, but then it gets
complicated having to deal with EL's /usr/lib{32,64} shenanigans.
[1]: https://github.com/repoforge/rpms/pull/250
Review: https://reviewboard.asterisk.org/r/2414/
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r384514 | mjordan | 2013-04-02 08:40:05 -0300 (Tue, 02 Apr 2013) | 5 lines
Make things work again
Sorry folks. ',' are still greater than '|'.
Thanks for playing along :-)
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r384518 | file | 2013-04-02 09:18:50 -0300 (Tue, 02 Apr 2013) | 2 lines
Pass the object type name to the configuration framework.
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r384546 | dlee | 2013-04-02 14:35:45 -0300 (Tue, 02 Apr 2013) | 15 lines
Fixed spurious rebuilds of func_version.
func_version.so was being rebuilt every time, because build.h was
changing every build, because of the cleantest dependency that was
added in r384410 to fix parallel make bugs.
Now build.h will only be created if it does not exist, which was the
original behavior of the Makefile.
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Merged revisions 384544 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 384545 from http://svn.asterisk.org/svn/asterisk/branches/11
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r384616 | rmudgett | 2013-04-03 13:01:51 -0300 (Wed, 03 Apr 2013) | 8 lines
astobj2: Fix rbtree duplicate handling.
OBJ_PARTIAL_KEY searching a rbtree did not find all possible matches if
the container did not accept duplicates.
Added matching node bias to indicate which matching node is being searched
for: first, last, any.
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r384642 | mjordan | 2013-04-03 14:17:33 -0300 (Wed, 03 Apr 2013) | 10 lines
Update documentation for CHANNEL function
Document that you can read/write the 'accountcode' and 'amaflags' on a channel.
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Merged revisions 384640 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r384696 | rmudgett | 2013-04-03 17:20:09 -0300 (Wed, 03 Apr 2013) | 26 lines
chan_dahdi: Add inband_on_proceeding compatibility option.
The new inband_on_proceeding option causes Asterisk to assume inband audio
may be present when a PROCEEDING message is received.
Q.931 Section 5.1.2 says the network cannot assume that the CPE side has
attached to the B channel at this time without explicitly sending the
progress indicator ie informing the CPE side to attach to the B channel
for audio. However, some non-compliant ISDN switches send a PROCEEDING
without the progress indicator ie indicating inband audio is available and
assume that the CPE device has connected the media path for listening to
ringback and other messages.
ASTERISK-17834 which causes this issue was dealing with a non-compliant
network switch.
(closes issue ASTERISK-21151)
Reported by: Gianluca Merlo
Tested by: rmudgett
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Merged revisions 384685 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 384689 from http://svn.asterisk.org/svn/asterisk/branches/11
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r384711 | rmudgett | 2013-04-03 17:27:11 -0300 (Wed, 03 Apr 2013) | 4 lines
chan_dahdi: Change inband_on_proceeding option default to no/disabled.
(issue ASTERISK-21151)
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r384760 | rmudgett | 2013-04-04 15:15:34 -0300 (Thu, 04 Apr 2013) | 2 lines
Separate some event struct definitions from instantiation.
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r384828 | elguero | 2013-04-05 17:41:27 -0300 (Fri, 05 Apr 2013) | 29 lines
Fix For Not Overriding The Default Settings In chan_sip
The initial report was that the "nat" setting in the [general] section was not
having any effect in overriding the default setting. Upon confirming that this
was happening and looking into what was causing this, it was discovered that
other default settings would not be overriden as well.
This patch works similar to what occurs in build_peer(). We create a temporary
ast_flags structure and using a mask, we override the default settings with
whatever is set in the [general] section.
In the bug report, the reporter who helped to test this patch noted that the
directmedia settings were being overriden properly as well as the nat settings.
This issue is also present in Asterisk 1.8 and a separate patch will be applied
to it.
(issue ASTERISK-21225)
Reported by: Alexandre Vezina
Tested by: Alexandre Vezina, Michael L. Young
Patches:
asterisk-21225-handle-options-default-prob_v4.diff
Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2385/
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Merged revisions 384827 from http://svn.asterisk.org/svn/asterisk/branches/11
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r384857 | file | 2013-04-06 13:00:20 -0300 (Sat, 06 Apr 2013) | 4 lines
Add a res_sorcery_astdb module which uses the astdb to persist objects.
Review: https://reviewboard.asterisk.org/r/2420/
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r384879 | dlee | 2013-04-08 10:27:45 -0300 (Mon, 08 Apr 2013) | 26 lines
Stasis application WebSocket support
This is the API that binds the Stasis dialplan application to external
Stasis applications. It also adds the beginnings of WebSocket
application support.
This module registers a dialplan function named Stasis, which is used
to put a channel into the named Stasis app. As a channel enters and
leaves the Stasis diaplan application, the Stasis app receives a
'stasis-start' and 'stasis-end' events.
Stasis apps register themselves using the stasis_app_register and
stasis_app_unregister functions. Messages are sent to an application
using stasis_app_send.
Finally, Stasis apps control channels through the use of the
stasis_app_control object, and the family of stasis_app_control_*
functions.
Other changes along for the ride are:
* An ast_frame_dtor function that's RAII_VAR safe
* Some common JSON encoders for name/number, timeval, and
context/extension/priority
Review: https://reviewboard.asterisk.org/r/2361/
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r384910 | mjordan | 2013-04-08 11:26:37 -0300 (Mon, 08 Apr 2013) | 17 lines
Add multi-channel Stasis messages; refactor Dial AMI events to Stasis
This patch does the following:
* A new Stasis payload has been defined for multi-channel messages. This
payload can store multiple ast_channel_snapshot objects along with a single
JSON blob. The payload object itself is opaque; the snapshots are stored
in a container keyed by roles. APIs have been provided to query for and
retrieve the snapshots from the payload object.
* The Dial AMI events have been refactored onto Stasis. This includes dial
messages in app_dial, as well as the core dialing framework. The AMI events
have been modified to send out a DialBegin/DialEnd events, as opposed to
the subevent type that was previously used.
* Stasis messages, types, and other objects related to channels have been
placed in their own file, stasis_channels. Unit tests for some of these
objects/messages have also been written.
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r384942 | mjordan | 2013-04-08 12:38:34 -0300 (Mon, 08 Apr 2013) | 9 lines
Don't attempt a websocket protocol removal if res_http_websocket isn't there
This patch sets the protocols container provided by res_http_websocket to NULL
when the module gets unloaded and adds the necessary checks when adding/
removing a websocket protocol. This prevents some FRACKing on an invalid
pointer to the disposed container if a module that uses res_http_websocket is
unloaded after it.
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r384989 | wdoekes | 2013-04-08 15:24:50 -0300 (Mon, 08 Apr 2013) | 4 lines
Clean up Makefile "warning" clutter when makeopts doesn't exist.
Review: https://reviewboard.asterisk.org/r/2304
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r385049 | newtonr | 2013-04-08 20:38:08 -0300 (Mon, 08 Apr 2013) | 10 lines
Modified the list of keys for the driver backends for sake of sample clarity
Added a line showing the mapping of "mysql" to res_config_mysql available in add-ons. We used "mysql" as an example driver key in the sample, but didn't show what module it mapped too. Also added a subtitle above the list of keys for driver backends.
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Merged revisions 385047 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 385048 from http://svn.asterisk.org/svn/asterisk/branches/11
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r385088 | russell | 2013-04-09 03:16:42 -0300 (Tue, 09 Apr 2013) | 10 lines
Add inheritance support to FEATURE()/FEATUREMAP().
The settings saved on the channel for FEATURE()/FEATUREMAP() were only
for that channel. This patch adds the ability to have these settings
inherited to child channels if you set FEATURE(inherit)=yes.
Closes issue ASTERISK-21306.
Review: https://reviewboard.asterisk.org/r/2415/
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r385116 | dlee | 2013-04-09 15:22:08 -0300 (Tue, 09 Apr 2013) | 6 lines
Backported app_stasis fix from stasis-http branch.
The hash and compare functions for the control container was reusing
the wrong ones, causing some problems. I fixed it, but in the wrong
branch. Oh well, it happens.
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r385142 | rmudgett | 2013-04-09 16:58:35 -0300 (Tue, 09 Apr 2013) | 4 lines
Rename struct feature_ds to struct feature_datastore.
Because "struct feature_ds *feature_ds" is not a good thing.
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r385174 | mjordan | 2013-04-10 11:07:27 -0300 (Wed, 10 Apr 2013) | 32 lines
Fix crash in chan_sip when a core initiated op occurs at the same time as a BYE
When a BYE request is processed in chan_sip, the current SIP dialog is detached
from its associated Asterisk channel structure. The tech_pvt pointer in the
channel object is set to NULL, and the dialog persists for an RFC mandated
period of time to handle re-transmits.
While this process occurs, the channel is locked (which is good).
Unfortunately, operations that are initiated externally have no way of knowing
that the channel they've just obtained (which is still valid) and that they are
attempting to lock is about to have its tech_pvt pointer removed. By the time
they obtain the channel lock and call the channel technology callback, the
tech_pvt is NULL.
This patch adds a few checks to some channel callbacks that make sure the
tech_pvt isn't NULL before using it. Prime offenders were the DTMF digit
callbacks, which would crash if AMI initiated a DTMF on the channel at the
same time as a BYE was received from the UA. This patch also adds checks on
sip_transfer (as AMI can also cause a callback into this function), as well
as sip_indicate (as lots of things can queue an indication onto a channel).
Review: https://reviewboard.asterisk.org/r/2434/
(closes issue ASTERISK-20225)
Reported by: Jeff Hoppe
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Merged revisions 385170 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 385173 from http://svn.asterisk.org/svn/asterisk/branches/11
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r385202 | mjordan | 2013-04-10 11:26:22 -0300 (Wed, 10 Apr 2013) | 25 lines
Use LDAP memory management functions instead of Asterisk's
When MALLOC_DEBUG is enabled with res_config_ldap, issues (munmap_chunk:
invalid pointer errors) can occur as the memory is being allocated with
Asterisk's wrappers around malloc/calloc/free/strdup, as opposed to the
LDAP library's wrappers.
This patch uses the LDAP library's wrappers where appropriate, so that
compiling with MALLOC_DEBUG doesn't cause more problems than it solves.
Note that the patch listed below was modified slightly for this commit
to account for some additional memory allocation/deallocations.
(closes issue ASTERISK-17386)
Reported by: John Covert
Tested by: Andrew Latham
patches:
issue18789-1.8-r316873.patch uploaded by seanbright (License 5060)
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Merged revisions 385190 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 385199 from http://svn.asterisk.org/svn/asterisk/branches/11
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r385236 | dlee | 2013-04-10 12:34:47 -0300 (Wed, 10 Apr 2013) | 5 lines
Fixed manager channelvars support.
For the events that have been ported to Stasis, this was broken in
r384910, when a couple of lines of code was lost in a merge.
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r385277 | rmudgett | 2013-04-10 20:03:30 -0300 (Wed, 10 Apr 2013) | 13 lines
* Fix unlocked accesses to feature_list. The feature_list is now also
protected by the features_lock.
* Made all calls to ast_find_call_feature() have the features_lock held.
* Fixed set_config_flags() to actually use find_group() to look for
feature groups in DYNAMIC_FEATURES. The code originally assumed all
feature groups were listed in DYNAMIC_FEATURES.
* Make everyone use ast_rdlock_call_features(),
ast_unlock_call_features(), and new ast_wrlock_call_features() instead of
directly calling the rwlock API on features_lock.
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r385278 | rmudgett | 2013-04-10 20:08:02 -0300 (Wed, 10 Apr 2013) | 1 line
Eliminated dial_features_destroy() since it is equivalent to ast_free_ptr()
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r385314 | rmudgett | 2013-04-11 13:53:21 -0300 (Thu, 11 Apr 2013) | 5 lines
Fix 'pri intense debug span' alias.
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Merged revisions 385313 from http://svn.asterisk.org/svn/asterisk/branches/11
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r385357 | qwell | 2013-04-11 17:00:46 -0300 (Thu, 11 Apr 2013) | 12 lines
Blocked revisions 385356
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Add dependency on libuuid, for res_rtp_asterisk
pjproject is what actually requires libuuid.
(closes issue ASTERISK-21125)
reported by Private Name
(Ed. note: Really? Private Name? I am rolling my eyes so hard right now.)
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r385406 | alecdavis | 2013-04-12 05:18:20 -0300 (Fri, 12 Apr 2013) | 24 lines
IAX2, prevent network thread starting before all helper threads are ready
On startup, it's possible for a frame to arrive before the processing threads were ready.
In iax2_process_thread() the first pass through falls into ast_cond_wait, should a frame arrive
before we are at ast_cond_wait, the signal will be ignored.
The result iax2_process_thread stays at ast_cond_wait forever, with deferred frames being queued.
Fix: When creating initial idle iax2_process_threads, wait for init_cond to be signalled
after each thread is started.
(issue ASTERISK-18827)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2427/
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Merged revisions 385402 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 385403 from http://svn.asterisk.org/svn/asterisk/branches/11
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r385431 | alecdavis | 2013-04-12 05:52:44 -0300 (Fri, 12 Apr 2013) | 17 lines
IAX2 defer_full_frames fail to get sent
Ensure iax2_process_thread is signalled when a deferred frame is queued to it.
(closes issue ASTERISK-18827)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2426/
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Merged revisions 385429 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 385430 from http://svn.asterisk.org/svn/asterisk/branches/11
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r385474 | elguero | 2013-04-12 12:06:09 -0300 (Fri, 12 Apr 2013) | 40 lines
Fix One-Way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX
When we reload Asterisk or chan_sip, the flags force_rport and comedia that are
turned on and off when using the auto_force_rport and auto_comedia nat settings
go back to the default setting off. These flags are turned on when needed or
off when not needed at the time that a peer registers, re-registers or initiates
a call. This would apply even when only the default global setting
"nat=auto_force_rport" is being used, which in this case would only affect the
force_rport flag.
Everything is good except for the following: The nat setting is set to
auto_force_rport and auto_comedia. We reload Asterisk and the peer's
registration has not expired. We load in the settings for the peer which turns
force_rport and comedia back to off. Since the peer has not re-registered or
placed a call yet, those flags remain off. We then initiate a call to the peer
from the PBX. The force_rport and comedia flags stay off. If NAT is involved,
we end up with one-way audio since we never checked to see if the peer is behind
NAT or not.
This patch does the following:
* Moves the checking of whether a peer is behind NAT into its own function
* Create a function to set the peer's NAT flags if they are using the auto_* NAT
settings
* Adds calls in sip_request_call() to these new functions in order to setup the
dialog according to the peer's settings
(closes issue ASTERISK-21374)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
asterisk-21374-auto-nat-outgoing-fix_v2.diff Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2421/
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Merged revisions 385473 from http://svn.asterisk.org/svn/asterisk/branches/11
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r385522 | kmoore | 2013-04-12 18:11:02 -0300 (Fri, 12 Apr 2013) | 5 lines
Expose channel snapshot manager blob generation
These functions are already used in one branch (jrose's parking branch)
and will soon be used in other branches as well.
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r385548 | qwell | 2013-04-12 18:48:10 -0300 (Fri, 12 Apr 2013) | 1 line
Fix documentation.
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r385573 | elguero | 2013-04-12 19:22:58 -0300 (Fri, 12 Apr 2013) | 36 lines
Fix app_voicemail Segfault And A Few Memory Leaks
The original report was that app_voicemail would crash. This was caused by
ast_config_load() returning CONFIG_STATUS_FILEINVALID but no checks being
performed for that return status. After adding the initial patch to fix this
issue, Jaco Kroon (jkroon) added some fixes to memory leaks he had discovered.
During review, Walter Doekes (wdoekes) suggested adding a helper function in
order to determine if we had a valid configuration or not.
This patch does the following:
[... 11631 lines stripped ...]
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