[svn-commits] file: branch file/gulp_fax r395104 - in /team/file/gulp_fax: ./ addons/ apps/...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jul 23 07:48:41 CDT 2013
Author: file
Date: Tue Jul 23 07:48:16 2013
New Revision: 395104
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=395104
Log:
Multiple revisions 393896-393897,393910,393919,393930,393968,393987,394004,394024,394037,394050,394065,394076,394089,394103,394147,394156,394158,394189,394203,394216,394232,394249,394263,394278,394290,394304-394305,394346,394370,394397,394402,394417,394442,394469-394471,394473,394489,394513,394530-394531,394552,394567,394583,394600,394623,394642,394686,394701,394715,394731,394744,394759,394776,394791,394793,394795,394808-394810,394825,394836,394846,394858,394870,394881,394894-394895,394898,394901,394939,394942,394982,395021,395034,395074,395088-395089,395102
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r393896 | rmudgett | 2013-07-09 16:38:00 -0300 (Tue, 09 Jul 2013) | 1 line
Fix some stasis doxygen comments.
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r393897 | mmichelson | 2013-07-09 17:07:21 -0300 (Tue, 09 Jul 2013) | 6 lines
Use correct function for getting bridged peer when doing direct media checks.
(closes issue ASTERISK-21947)
reported by Matt Jordan
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r393910 | rmudgett | 2013-07-09 18:06:21 -0300 (Tue, 09 Jul 2013) | 1 line
Fix printf NULL string (null) substituion for NULL config framework default.
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r393919 | qwell | 2013-07-09 18:40:38 -0300 (Tue, 09 Jul 2013) | 7 lines
Make SCOPED_LOCK use RAII_VAR.
This fixes an issue with requiring SCOPED_LOCK to be the last variable
declaration and removes duplicate code in the process.
Review: https://reviewboard.asterisk.org/r/2665/
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r393930 | russell | 2013-07-09 22:56:15 -0300 (Tue, 09 Jul 2013) | 26 lines
astobj2-ify the SLA code
The SLA code within app_meetme was written before asotbj2 had been
merged into Asterisk. Worse, support for reloads did not exist at first
and was added later as a bolt-on feature. I knew at the time that
reloading was not safe at all while SLA was in use, so the reload would
be queued up to execute when the system was idle. Unfortunately, this
approach was still prone to errors beyond the fact that this was the
only place in Asterisk where configuration was not reloaded
instantly when requested.
This patch converts various SLA objects to be reference counted objects
using astobj2. This allows reloads to be processed while the system is
in use. The code ensures that the objects will not disappear while one
of the other threads is using them. However, they will be immediately
removed from the global trunk and station containers so no new calls
will use them if removed from configuration.
Review: https://reviewboard.asterisk.org/r/2581/
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Merged revisions 393928 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 393929 from http://svn.asterisk.org/svn/asterisk/branches/11
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r393968 | dlee | 2013-07-10 10:50:48 -0300 (Wed, 10 Jul 2013) | 1 line
Corrected api-docs for channel variables
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r393987 | dlee | 2013-07-10 14:13:21 -0300 (Wed, 10 Jul 2013) | 1 line
Document the 400 error response for originate
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r394004 | file | 2013-07-10 17:02:59 -0300 (Wed, 10 Jul 2013) | 5 lines
Handle outbound registration failures that do not occur as a result of a real response.
(closes issue ASTERISK-22064)
Reported by: Rusty Newton
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r394024 | kharwell | 2013-07-10 19:26:13 -0300 (Wed, 10 Jul 2013) | 23 lines
PSJIP - sip.conf to res_sip.conf script
** This script is in no way finished.
Started the initial "cut" at converting a sip.conf file to a res_sip.conf file.
Hopefully the bulk of the framework is in place and only a few minor adjustments
need to be made when an option mapping is added that "doesn't fit". This script
and supporting files should be executable against python version 2.5.
An OrderedDict class (backported from a newer version of python) is included.
A MultiOrderedDict class is implemented so options, when added, should be able
to be added in order and allowed to have multiple values.
Currently the scripts supports the majority of endpoint options found in
res_sip.conf. Support has also been added for Aor(s) and the ACL/security
sections. Inside the sip_to_res_sip.py file one can see a list of options
that still need to be mapped.
Also items that still need to be done: templates, includes, parsing '=>'
delimiter. Note that some code is hopefully in place already to support
templates (e.g. lookup/retrieving defaults from them). However, the
parsing of and adding of the section needs to be done.
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r394037 | dlee | 2013-07-10 23:02:48 -0300 (Wed, 10 Jul 2013) | 1 line
Fixed some CEL test crashes
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r394050 | dlee | 2013-07-11 01:34:49 -0300 (Thu, 11 Jul 2013) | 9 lines
test_voicemail_api: fix warning found by gcc-4.8
The voicemail_api test had code like strncmp(a, b, sizeof(a)), but a was a
char pointer, instead of a literal or char array. This meant that sizeof was
the size of the pointer, not the length of the string.
Since the string is in a stringfield and should be null terminated, I just
changed it to a plain strcmp.
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r394065 | dlee | 2013-07-11 10:56:26 -0300 (Thu, 11 Jul 2013) | 1 line
Apply defaults to ari.conf's general section
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r394076 | dlee | 2013-07-11 11:39:55 -0300 (Thu, 11 Jul 2013) | 11 lines
Change ARI user config to use a type field
When I initially wrote the configuration support for ARI users, I
determined the section type by a category prefix (i.e., [user-admin]).
This is neither idiomatic Asterisk configuration, nor is it really
that user friendly. This patch replaces the category prefix with a
type field in the section, which is much cleaner.
Review: https://reviewboard.asterisk.org/r/2664/
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r394089 | dlee | 2013-07-11 12:37:51 -0300 (Thu, 11 Jul 2013) | 12 lines
Correct test_cel cleanup.
When I corrected the CEL test crash in r394037, I didn't quite pay attention
to how the globals and locals were being shuffled around in the cleanup
callback. I removed the nulling of the global variables, which caused them
to be double cleaned.
This patch puts the global nulling code back (since the vars are cleaned up
by RAII_VARs), and removes the explicit ao2_cleanup() (since they were no-ops,
because the variables had just been nulled).
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r394103 | file | 2013-07-11 13:23:41 -0300 (Thu, 11 Jul 2013) | 2 lines
Tweak the subscription failure warning message to include endpoint name and context.
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r394147 | wedhorn | 2013-07-11 17:17:53 -0300 (Thu, 11 Jul 2013) | 16 lines
Refactor and cleanup of skinny session handling.
Major changes are to pull all packet reading functions into skinny_session
and move timeout handling to scheduling arrangements. Thread cancelling is
now undertaken directly rather than waiting for the read to timeout
(cleanup is popped on thread cancel). Also added some keepalive timings in
debugging messages.
Keepalive timeout has been increased from 1.1 by keepalive to 3 times
keepalive. This seems to align (after keepalives stabilise) with when
devices reset after not receiving keepalives. Probably needs more work,
especially around the first and/or second keepalives that vary
significantly by device and firmware version.
Review: https://reviewboard.asterisk.org/r/2611/
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r394156 | dlee | 2013-07-11 17:59:10 -0300 (Thu, 11 Jul 2013) | 4 lines
Fixed chan_skinny for systems were pthread_t isn't an int.
I'm looking at you, OS X.
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r394158 | rmudgett | 2013-07-11 18:01:09 -0300 (Thu, 11 Jul 2013) | 1 line
Fix bridge tech write callback parameter name.
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r394189 | qwell | 2013-07-12 14:52:52 -0300 (Fri, 12 Jul 2013) | 8 lines
ARI: Add support for Cross-Origin Resource Sharing (CORS), origin headers
This rejects requests from any unknown origins.
(closes issue ASTERISK-21278)
Review: https://reviewboard.asterisk.org/r/2667/
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r394203 | dlee | 2013-07-12 15:23:39 -0300 (Fri, 12 Jul 2013) | 18 lines
Fixed intermittent crash when loading test_json.so
The JSON test attempted an overly clever use of RAII_VAR to run code
at the beginning and end of each test, in order to validate that no
JSON objects were leaked during the test.
The problem is that the validation code would run during the initial
load, when the tests were initialized. This happens during startup,
when other parts of the system might actively be allocating and
freeing JSON objects.
This patch changes the RAII_VAR to use the new
ast_test_register_{init,cleanup} functions to run the validations
properly.
(closes issue ASTERISK-21978)
Review: https://reviewboard.asterisk.org/r/2669/
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r394216 | qwell | 2013-07-12 16:35:08 -0300 (Fri, 12 Jul 2013) | 2 lines
Fix a compiler warning.
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r394232 | mmichelson | 2013-07-12 18:01:51 -0300 (Fri, 12 Jul 2013) | 11 lines
Prevent potential race condition in multiparty basic bridges.
For more details about the race condition see the linked review
at the bottom of this commit
(closes issue ASTERISK-21882)
Reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2663
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r394249 | file | 2013-07-12 18:42:53 -0300 (Fri, 12 Jul 2013) | 6 lines
Add support to the bridging core for performing COLP updates when channels join a 2 party bridge.
(closes issue ASTERISK-21829)
Review: https://reviewboard.asterisk.org/r/2636/
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r394263 | moy | 2013-07-12 19:35:50 -0300 (Fri, 12 Jul 2013) | 21 lines
Fix a longstanding issue with MFC-R2 configuration that prevented users
from mixing different variants or general MFC-R2 settings within the same E1 line.
Most users do not have a problem with this since MFC-R2 lines are usually fractional E1s, or
the whole E1 has the same country variant and R2 settings.
In Venezuela however is common to have inbound MFC-R2 and outbound DTMF-R2 within the same E1.
This fix now properly parses the chan_dahdi.conf file to generate a new openr2 context every
time a new channel => section is found and the configuration was changed.
(closes issue ASTERISK-21117)
Reported by: Rafael Angulo
Related Elastix issue: http://bugs.elastix.org/view.php?id=1612
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Merged revisions 394106 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 394173 from http://svn.asterisk.org/svn/asterisk/branches/11
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r394278 | mjordan | 2013-07-13 19:14:04 -0300 (Sat, 13 Jul 2013) | 5 lines
Pretty up a debug message if the referred-by-uri isn't available
Instead of formatting a NULL pointer into a "%s" format string (which is
usually not a good thing to do), we instead print "Unknown".
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r394290 | mjordan | 2013-07-13 20:28:23 -0300 (Sat, 13 Jul 2013) | 27 lines
Fix FRACK message from external redirects; handle outbound channels better
This patch does the following:
* It simplifies the Dial handling in CDRs. As a rule, the caller in a dial
relationship is always the Party A. There was some logic present in the
handling of the dial message that could, conceivably, pick the caller
as Party A for the beginning of the dial and the peer as Party A for the
end of the dial. This shouldn't have happened if the code in the bridging
framework was doing its job; however, that was broken and it led to the
FRACK. As it is, this code was overly ocmplex and not needed: the caller,
if present, should always be Party A. Period.
* It properly checks to see if a channel will continue on in the dialplan.
ast_check_hangup - much like cake at the end - is a lie. It will tell
you that you are hungup when you are not. Do not believe it.
I would make this function tell the truth, but I'm nervous that we've been
depending on it sitting on its throne of lies for far too long, and it would
probably break lots of things. So I'm just checking the "internal" soft
hangup flags, like everyone else.
(closes issue ASTERISK-22060)
Reported by: Mark Michelson
(issue ASTERISK-21831)
Reported by: Matt Jordan
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r394304 | mjordan | 2013-07-13 23:00:02 -0300 (Sat, 13 Jul 2013) | 19 lines
Clarify documentation for function PASSTHRU
It is not apparent to the average user that the PASSTHRU function should not
be passed as ${PASSTHRU(string)} but just as PASSTHRU(string) to functions
which take a variable name and not its contents.
This patch clarifies the behavior in the documentation and provides an example.
(closes issue ASTERISK-21717)
Reported by: Richard Miller
patches:
func_strings.diff uploaded by Richard Miller (license 5685)
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Merged revisions 394302 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 394303 from http://svn.asterisk.org/svn/asterisk/branches/11
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r394305 | mjordan | 2013-07-13 23:05:04 -0300 (Sat, 13 Jul 2013) | 11 lines
Remove redundant code in dns.c
Peter J Philipp pointed out that there are two checks that ensure that len is
not less than 0. If len is less than 0, the function returns. Having both of
them is clearly redundant.
This removes the second and attempts to clarify (slightly) the error condition.
(closes issue ASTERISK-21772)
Reported by: Peter J Philipp
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r394346 | mjordan | 2013-07-13 23:41:43 -0300 (Sat, 13 Jul 2013) | 13 lines
Provide error message for QUEUE_MEMBER when member is not in queue
When QUEUE_MEMBER is used and the member specified is not in the queue,
Asterisk provides an ERROR message that indicates that the option specified
is not valid. This patch now properly displays an ERROR message that the
member is not in the queue if an interface is specified.
(closes issue ASTERISK-21980)
Reported by: Avraam David
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Merged revisions 394345 from http://svn.asterisk.org/svn/asterisk/branches/11
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r394370 | file | 2013-07-15 10:43:37 -0300 (Mon, 15 Jul 2013) | 2 lines
Remove some callbacks and functions which are not needed.
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r394397 | dlee | 2013-07-15 18:22:12 -0300 (Mon, 15 Jul 2013) | 1 line
Document the ari.conf allowed_origins setting
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r394402 | mmichelson | 2013-07-15 19:05:57 -0300 (Mon, 15 Jul 2013) | 3 lines
Remove misleading documentation for channel snapshot creation.
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r394417 | rmudgett | 2013-07-15 20:20:55 -0300 (Mon, 15 Jul 2013) | 60 lines
Replace chan_agent with app_agent_pool.
The ill conceived chan_agent is no more. It is now replaced by
app_agent_pool.
Agents login using the AgentLogin() application as before. The
AgentLogin() application no longer does any authentication.
Authentication is now the responsibility of the dialplan. (Besides, the
authentication done by chan_agent did not match what the voice prompts
asked for.)
Sample extensions.conf
[login]
; Sample agent 1001 login
; Set COLP for in between calls so the agent does not see the last caller COLP.
exten => 1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>)
; Give the agent DTMF transfer and disconnect features when connected to a caller.
same => n,Set(CHANNEL(dtmf-features)=TX)
same => n,AgentLogin(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()
[caller]
; Sample caller direct connect to agent 1001
exten => 800,1,AgentRequest(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()
; Sample caller going through a Queue to agent 1001
exten => 900,1,Queue(agent_q)
same => n,Hangup()
Sample queues.conf
[agent_q]
member => Local/800 at caller,,SuperAgent,Agent:1001
Under the hood operation overview:
1) Logged in agents wait for callers in an agents holding bridge.
2) Caller requests an agent using AgentRequest()
3) A basic bridge is created, the agent is notified, and caller joins the
basic bridge to wait for the agent.
4) The agent is either automatically connected to the caller or must ack
the call to connect.
5) The agent is moved from the agents holding bridge to the basic bridge.
6) The agent and caller talk.
7) The connection is ended by either party.
8) The agent goes back to the agents holding bridge.
To avoid some locking issues with the agent holding bridge, I needed to
make some changes to the after bridge callback support. The after bridge
callback is now a list of requested callbacks with the last to be added
the only active callback. The after bridge callback for failed callbacks
will always happen in the channel thread when the channel leaves the
bridging system or is destroyed.
(closes issue ASTERISK-21554)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2657/
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r394442 | dlee | 2013-07-16 12:30:09 -0300 (Tue, 16 Jul 2013) | 1 line
Fixed null dereference when WebSocket protocol is omitted
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r394469 | mjordan | 2013-07-16 15:22:07 -0300 (Tue, 16 Jul 2013) | 9 lines
Re-order cleanup
This patch attempts to fix some possible race conditions in shutdown of the
CDR engine. It:
* Adds a cleanup handler to only unsubscribe and join on stasis messages during
graceful shutdown. The cleanup handler should execute before the regular atexit
handler, as we want to unsubscribe for any further messages before dispatching
the CDRs.
* The CDRs are now locked when we dispatch them on shutdown.
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r394470 | rmudgett | 2013-07-16 15:48:49 -0300 (Tue, 16 Jul 2013) | 1 line
Simplify bridge_simple chan join code.
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r394471 | rmudgett | 2013-07-16 15:49:26 -0300 (Tue, 16 Jul 2013) | 1 line
Remove some dead code dealing with old bridging method.
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r394473 | mmichelson | 2013-07-16 16:13:04 -0300 (Tue, 16 Jul 2013) | 9 lines
Prevent crash from trying to end a session in an invalid way.
This ensures that code that was only meant to be run on a reinvite failure
only runs on a reinvite failure.
(closes issue ASTERISK-22061)
reported by Rusty Newton
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r394489 | rmudgett | 2013-07-16 17:00:25 -0300 (Tue, 16 Jul 2013) | 1 line
chan_gulp: Fix gulp_indicate() handling of AST_CONTROL_PVT_CAUSE_CODE.
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r394513 | dlee | 2013-07-16 18:44:12 -0300 (Tue, 16 Jul 2013) | 1 line
Debug logging to help with WebSocket connection problems
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r394530 | mjordan | 2013-07-16 19:25:33 -0300 (Tue, 16 Jul 2013) | 7 lines
Re-order handlers in CEL to ensure that HANGUP events happen after APP_END
When a channel is hungup, both an APP_END event and a HANGUP event can be
fired. To ensure that HANGUP events occur after APP_END events, the method
callbacks for the APP_END event should be processed prior to the callbacks
for the HANGUP event.
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r394531 | mjordan | 2013-07-16 19:33:27 -0300 (Tue, 16 Jul 2013) | 13 lines
Add 'kick all' capability to ConfBridge CLI command
This patch adds the ability to kick all users out of a conference from the
ConfBridge kick CLI command. It is invoked by passing 'all' as the channel
parameter to the CLI command, i.e., "confbridge kick <conf> all".
Note that this patch was modified slightly to conform to trunk.
(closes issue ASTERISK-21827)
Reported by: dorianlogan
patches:
kickall-patch_v2.diff uploaded by dorianlogan (License 6504)
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r394552 | tzafrir | 2013-07-17 13:56:14 -0300 (Wed, 17 Jul 2013) | 20 lines
handle DAHDI_EVENT_REMOVED on a pri D-Channel
When a DAHDI device is removed at run-time it sends the event
DAHDI_EVENT_REMOVED on each channel. This is intended to signal the
userspace program to close the respective file handle, as the driver of
the device will need all of them closed to properly clean-up.
This event has long since been handled in chan_dahdi (chan_zap at the
time). However the event that is sent on a D-Channel of a "PRI" (ISDN)
span simply gets ignored.
This commit adds handling for closing the file descriptor (and shutting
down the span, while we're at it).
It also adds a CLI command 'pri destroy span <N>' to destroy the span
and its DAHDI channels.
Review: https://reviewboard.asterisk.org/r/726/
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r394567 | tzafrir | 2013-07-17 14:49:54 -0300 (Wed, 17 Jul 2013) | 2 lines
Left over spacing issues of review 726.
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r394583 | jrose | 2013-07-17 15:26:19 -0300 (Wed, 17 Jul 2013) | 2 lines
app_confbridge: Eliminate a reference leak for confbridge announcer channels
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r394600 | rmudgett | 2013-07-17 16:27:26 -0300 (Wed, 17 Jul 2013) | 1 line
Remove some completed and no longer relevant BUGBUG notes.
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r394623 | rmudgett | 2013-07-17 19:30:28 -0300 (Wed, 17 Jul 2013) | 11 lines
Change ast_hangup() to return void and be NULL safe.
Since ast_hangup() is effectively a channel destructor, it should be a
void function.
* Make the few silly callers checking the return value no longer do so.
Only the CDR and CEL unit tests checked the return value.
* Make all callers take advantage of the NULL safe change and remove the
NULL check before the call.
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r394642 | elguero | 2013-07-18 09:54:50 -0300 (Thu, 18 Jul 2013) | 25 lines
Properly indicate failure to open an audio stream in res_agi
If there is an error streaming an audio file, the current return status makes it
difficult for an AGI script to determine that there was an error with the audio
file.
This patches changes the result to return -1 and the function returns
RESULT_FAILURE instead of RESULT_SUCCESS. From looking at other parts of
res_agi, this would appear to be the proper way to handle an error.
(closes issue ASTERISK-21903)
Reported by: Ariel Wainer
Tested by: Ariel Wainer
Patches:
asterisk-21903-return-stream-res_1.8.diff
by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2625/
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Merged revisions 394640 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 394641 from http://svn.asterisk.org/svn/asterisk/branches/11
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r394686 | dlee | 2013-07-18 11:08:21 -0300 (Thu, 18 Jul 2013) | 24 lines
Fix caching topic shutdown assertions
The recent changes to update stasis_cache_topics directly from the
publisher thread uncovered a race condition, which was causing asserts
in the /stasis/core tests.
If the caching topic's subscription is the last reference to the
caching topic, it will destroy the caching topic after the final
message has been processed. When dispatching to a different thread,
this usually gave the unsubscribe enough time to finish before
destruction happened. Now, however, it consistently destroys before
unsubscription is complete.
This patch adds an extra reference to the caching topic, to hold it
for the duration of the unsubscription.
This patch also removes an extra unref that was happening when the
final message was received by the caching topic. It was put there
because of an extra ref that was put into the caching topic's
constructor. Both have been removed, which makes the destructor a bit
less confusing.
Review: https://reviewboard.asterisk.org/r/2675/
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r394701 | mjordan | 2013-07-18 11:50:56 -0300 (Thu, 18 Jul 2013) | 10 lines
Tweak debug statements
This patch does two things:
1. It moves the debug statement that shows the HTTP sub-protocols being
compared after the string length calculation such that it shows the correct
string length in the output
2. It adds some additional debug that displays when it matches on a
sub-protocol and when it fails
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r394715 | qwell | 2013-07-18 13:03:12 -0300 (Thu, 18 Jul 2013) | 8 lines
ARI: Add support for suppressing media streams.
Also convert res_mutestream to use the core feature behind this.
(closes issue ASTERISK-21618)
Review: https://reviewboard.asterisk.org/r/2652/
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r394731 | jrose | 2013-07-18 13:49:44 -0300 (Thu, 18 Jul 2013) | 14 lines
bridge_holding/app_bridgewait: Add new entertainment options
This patch adds more entertainment options to holding bridges and the
bridge_wait application. Also, holding bridges will now use music on
hold as the default entertainment option instead of none. The
parameters for app_bridgewait have changed to (role, options) from
the previous (options) and the options themselves have changed as
well (entertainment options are now contained in an enumerator, role
specification is handled by the role parameter, etc)
(closes issue ASTERISK-21923)
Reported by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/2679/
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r394744 | dlee | 2013-07-18 15:05:07 -0300 (Thu, 18 Jul 2013) | 1 line
Fixed null dereference when WebSocket subprotocol isn't specified
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r394759 | mmichelson | 2013-07-18 16:25:51 -0300 (Thu, 18 Jul 2013) | 11 lines
Add a bunch of options from sip.conf to res_sip.conf
For a complete list of the options added, see the review linked
at the bottom of this commit message.
(closes issue ASTERISK-21506)
reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2671
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r394776 | rmudgett | 2013-07-18 19:01:42 -0300 (Thu, 18 Jul 2013) | 1 line
Fixup doxygen on ast_hangup().
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r394791 | rmudgett | 2013-07-19 14:55:49 -0300 (Fri, 19 Jul 2013) | 10 lines
Made audiohooks, framehooks, and monitor prevent local channel optimization.
Audiohooks, framehooks, and monitor represent state on a local channel
that will go away if it is optimized out.
(closes issue ASTERISK-21954)
Reported by: rmudgett
Review: https://reviewboard.asterisk.org/r/2685/
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r394793 | qwell | 2013-07-19 15:00:35 -0300 (Fri, 19 Jul 2013) | 6 lines
Convert CCSS manager events to stasis.
(closes issue ASTERISK-21473)
Review: https://reviewboard.asterisk.org/r/2682/
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r394795 | kmoore | 2013-07-19 15:09:43 -0300 (Fri, 19 Jul 2013) | 7 lines
Fix crash when using temporary peers
Temporary peers do not have an associated Stasis endpoint and quite a
bit of code in chan_sip assumes that all peers have a Stasis endpoint.
All endpoint accesses in chan_sip are now wrapped in an endpoint
NULL-check.
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r394808 | kmoore | 2013-07-19 16:23:39 -0300 (Fri, 19 Jul 2013) | 11 lines
Filter channels used as internal mechanisms
This adds new flags to the channel tech properties that flag it as
different types of implementation detail used exclusively to provide a
feature. Examples of channels that would have these flags include the
announcement and recording channels used by confbridge which are the
only two marked as such by this patch.
Review: https://reviewboard.asterisk.org/r/2633/
(closes issue ASTERISK-21873)
........
r394809 | jrose | 2013-07-19 16:35:21 -0300 (Fri, 19 Jul 2013) | 17 lines
ARI: Bridge Playback, Bridge Record
Adds a new channel driver for creating channels for specific purposes
in bridges, primarily to act as either recorders or announcers. Adds
ARI commands for playing announcements to ever participant in a bridge
as well as for recording a bridge. This patch also includes some
documentation/reponse fixes to related ARI models such as playback
controls.
(closes issue ASTERISK-21592)
Reported by: Matt Jordan
(closes issue ASTERISK-21593)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2670/
........
r394810 | jrose | 2013-07-19 16:40:27 -0300 (Fri, 19 Jul 2013) | 6 lines
ARI: MOH start and stop for a channel
(issue ASTERISK-21974)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2680/
........
r394825 | rmudgett | 2013-07-19 19:47:10 -0300 (Fri, 19 Jul 2013) | 2 lines
Extract a repeated test into ast_channel_has_audio_frame_or_monitor().
........
r394836 | rmudgett | 2013-07-19 20:30:10 -0300 (Fri, 19 Jul 2013) | 7 lines
Minor optimizations.
* Made ast_audiohook_detach_list() and ast_audiohook_write_list_empty()
NULL tolerant.
* Made ast_audiohook_detach_list() return void since it is a destructor.
........
r394846 | rmudgett | 2013-07-19 22:11:38 -0300 (Fri, 19 Jul 2013) | 10 lines
Regroup the ao2 search_flags.
Moved the OBJ_POINTER, OBJ_KEY, and OBJ_PARTIAL_KEY flags together into a
field and renamed them to OBJ_SEARCH_OBJECT, OBJ_SEARCH_KEY, and
OBJ_SEARCH_PARTIAL_KEY respectively. The values were selected to keep
existing code compiling and working until the codebase can be changed to
stop using these values as bit flags and use them as an enum field.
The old names are defined to the new names for backward compatibility.
........
r394858 | kmoore | 2013-07-20 10:10:22 -0300 (Sat, 20 Jul 2013) | 16 lines
Add transfer support to CEL
This adds CEL support for blind and attended transfers and call pickup.
During the course of adding this functionality I noticed that
CONF_ENTER, CONF_EXIT, and BRIDGE_TO_CONF events are particularly
useless without a bridge identifier, so I added that as well.
This adds tests for blind transfers, several types of attended
transfers, and call pickup.
The extra field in CEL records now consists of a JSON blob whose fields
are defined on a per-event basis.
Review: https://reviewboard.asterisk.org/r/2658/
(closes issue ASTERISK-21565)
........
r394870 | kmoore | 2013-07-20 10:25:05 -0300 (Sat, 20 Jul 2013) | 9 lines
Add CEL local optimization record type
This adds a new CEL event type, AST_CEL_LOCAL_OPTIMIZE, to represent
local channel optimizations. Local channel optimizations were one of
several things conveyed by the now defunct BRIDGE_UPDATE event type.
This also adds a unit test to test generation of this new CEL event.
Review: https://reviewboard.asterisk.org/r/2676/
........
r394881 | mjordan | 2013-07-20 23:11:49 -0300 (Sat, 20 Jul 2013) | 11 lines
Allow setting allowmultiplelogin on an account basis
This patch modifies manager to allow the allowmultiplelogin setting to be set
on an account by account basis. When set in the general context, it will act
as the default for the defined accounts. Setting it in the account will
override the general setting.
(closes issue ASTERISK-21324)
Reported by: vldmr
patches:
asterisk-manager-per-user-allowmultiplelogin.patch uploaded by vldmr (License 6487)
........
r394894 | mjordan | 2013-07-20 23:26:31 -0300 (Sat, 20 Jul 2013) | 14 lines
Document connectedline parameter for chan_iax2
The connectedline parameter for a chan_iax2 peer was undocumented. This patch
documents the options in the sample configuration file.
(closes issue ASTERISK-21953)
Reported by: Birger "WIMPy" Harzenetter
........
Merged revisions 394886 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 394890 from http://svn.asterisk.org/svn/asterisk/branches/11
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r394895 | mjordan | 2013-07-20 23:30:58 -0300 (Sat, 20 Jul 2013) | 11 lines
Add additional control frame types to the IAX2 parser for debug messages
This patch adds some of the more recent control frame types to the IAX2
parser. When IAX2 debugging is enabled, it will now show more of the control
frame types.
(closes issue ASTERISK-22120)
Reported by: Birger "WIMPy" Harzenetter
patches:
iaxcmds.diff uploaded by wimpy
........
r394898 | mjordan | 2013-07-20 23:39:37 -0300 (Sat, 20 Jul 2013) | 17 lines
Update PostgreSQL realtime scripts with schema for queue_log table
This patch updates the realtime SQL scripts with an entry that will create the
queue_log table. This brings the PostgreSQL scripts inline with the MySQL
scripts, with respect to what tables they will create.
(closes issue ASTERISK-21021)
Reported by: Eugene
patches:
queue_log.sql uploaded by varnav (license 6360)
........
Merged revisions 394896 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 394897 from http://svn.asterisk.org/svn/asterisk/branches/11
........
r394901 | mjordan | 2013-07-21 00:10:41 -0300 (Sun, 21 Jul 2013) | 26 lines
Tolerate presence of RFC2965 Cookie2 header by ignoring it
This patch modifies parsing of cookies in Asterisk's http server by doing an
explicit comparison of the "Cookie" header instead of looking at the first
6 characters to determine if the header is a cookie header. This avoids
parsing "Cookie2" headers and overwriting the previously parsed "Cookie"
header.
Note that we probably should be appending the cookies in each "Cookie"
header to the parsed results; however, while clients can send multiple
cookie headers they never really do. While this patch doesn't improve
Asterisk's behavior in that regard, it shouldn't make it any worse either.
Note that the solution in this patch was pointed out on the issue by the
issue reporter, Stuart Henderson.
(closes issue ASTERISK-21789)
Reported by: Stuart Henderson
Tested by: mjordan, Stuart Henderson
........
Merged revisions 394899 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 394900 from http://svn.asterisk.org/svn/asterisk/branches/11
........
r394939 | mjordan | 2013-07-21 15:12:00 -0300 (Sun, 21 Jul 2013) | 17 lines
Always install safe_asterisk; add configuration file support
This patch modifies the behavior of safe_asterisk in two ways:
(1) It modifies the Asterisk Makefile such that safe_asterisk is always
installed on a 'make install'. This was done as bugfixes in the
safe_asterisk script were not applied in previous version of Asterisk
without first removing the old version of the script.
(2) In order to keep a newly installed version of safe_asterisk from impacting
local modifications, a new config file - safe_asterisk.conf.sample - has
been provided. Settings that were previously modified in safe_asterisk can
be set there instead.
(closes issue ASTERISK-21965)
Reported by: Jeremy Kister
patches:
safe_asterisk.patch uploaded by jkister (License 6232)
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r394942 | mjordan | 2013-07-21 15:17:17 -0300 (Sun, 21 Jul 2013) | 19 lines
Provide proper ring tone in indications.conf for Malaysia
The ring tone provided in the sample indications.conf was incorrect. This patch
modifies the sample ring tone to be what it should:
ring = 425/400,0/200,425/400,0/2000
This brings it in line with the tone definition in DAHDI 2.7.0. (zonedata.c)
(closes issue ASTERISK-21997)
Reported by: Filip Jenicek
patches:
malaysia_ring.patch uploaded by phill (License 6277)
........
Merged revisions 394940 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 394941 from http://svn.asterisk.org/svn/asterisk/branches/11
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r394982 | mjordan | 2013-07-21 19:09:23 -0300 (Sun, 21 Jul 2013) | 19 lines
Clean up documentation
This patch cleans up documentation in func_channel for the following items:
* rtpsource
* secure_signaling
* secure_media
* various OOH323 parameters
(closes issue ASTERISK-20969)
Reported by: snuffy
patches:
func_chan-update.diff uploaded by snuffy (License 5024)
........
Merged revisions 394980 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 394981 from http://svn.asterisk.org/svn/asterisk/branches/11
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r395021 | mjordan | 2013-07-21 20:13:29 -0300 (Sun, 21 Jul 2013) | 14 lines
Blocked revisions 395020
........
Add an upgrade note for libuuid dependency; remove note in CHANGES
This patch notes that libuuid is now a dependency for res_rtp_asterisk; this
was introduced in between 11.4.0 and 11.5.0 to resolve a dependency for
pjproject, which res_rtp_asterisk uses for ICE/STUN/TURN support.
It also removes a conflicting note from CHANGES. While support for playing
prompts to the first participant was added for app_queue, it was disabled
by default and an option added to enable it. That was properly noted in the
UPGRADE.txt file.
........
r395034 | mjordan | 2013-07-22 10:52:10 -0300 (Mon, 22 Jul 2013) | 11 lines
Update copyright year to 2013 in asterisk.c; some whitespace fixes
(closes issue ASTERISK-22179)
Reported by: Malcolm Davenport
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Merged revisions 395032 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 395033 from http://svn.asterisk.org/svn/asterisk/branches/11
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r395074 | kmoore | 2013-07-22 17:42:28 -0300 (Mon, 22 Jul 2013) | 1 line
Make the CEL blind transfer test pass consistently
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r395088 | rmudgett | 2013-07-22 21:02:04 -0300 (Mon, 22 Jul 2013) | 1 line
Remove some BUGBUG notes that have been handled.
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r395089 | mjordan | 2013-07-22 21:16:59 -0300 (Mon, 22 Jul 2013) | 3 lines
Fix unbalanced lock when serializing CDR variables
I'm only surprised that this didn't cause larger problems.
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r395102 | file | 2013-07-23 09:27:03 -0300 (Tue, 23 Jul 2013) | 7 lines
Expose the chan_pjsip implementation pvt and session in a defined manner.
This allows modules outside of chan_pjsip itself to get the session given
only an Asterisk channel.
Review: https://reviewboard.asterisk.org/r/2674/
........
Merged revisions 393896-393897,393910,393919,393930,393968,393987,394004,394024,394037,394050,394065,394076,394089,394103,394147,394156,394158,394189,394203,394216,394232,394249,394263,394278,394290,394304-394305,394346,394370,394397,394402,394417,394442,394469-394471,394473,394489,394513,394530-394531,394552,394567,394583,394600,394623,394642,394686,394701,394715,394731,394744,394759,394776,394791,394793,394795,394808-394810,394825,394836,394846,394858,394870,394881,394894-394895,394898,394901,394939,394942,394982,395021,395034,395074,395088-395089,395102 from http://svn.asterisk.org/svn/asterisk/trunk
Added:
team/file/gulp_fax/apps/app_agent_pool.c
- copied unchanged from r395102, trunk/apps/app_agent_pool.c
team/file/gulp_fax/channels/chan_bridge_media.c
- copied unchanged from r395102, trunk/channels/chan_bridge_media.c
team/file/gulp_fax/configs/safe_asterisk.conf.sample
- copied unchanged from r395102, trunk/configs/safe_asterisk.conf.sample
team/file/gulp_fax/contrib/scripts/sip_to_res_sip/
- copied from r395102, trunk/contrib/scripts/sip_to_res_sip/
team/file/gulp_fax/res/res_sip/config_global.c
- copied unchanged from r395102, trunk/res/res_sip/config_global.c
team/file/gulp_fax/res/res_sip/config_system.c
- copied unchanged from r395102, trunk/res/res_sip/config_system.c
team/file/gulp_fax/res/res_sip/sip_global_headers.c
- copied unchanged from r395102, trunk/res/res_sip/sip_global_headers.c
Removed:
team/file/gulp_fax/channels/chan_agent.c
Modified:
team/file/gulp_fax/ (props changed)
team/file/gulp_fax/CHANGES
team/file/gulp_fax/Makefile
team/file/gulp_fax/UPGRADE.txt
team/file/gulp_fax/addons/chan_mobile.c
team/file/gulp_fax/apps/app_bridgewait.c
team/file/gulp_fax/apps/app_celgenuserevent.c
team/file/gulp_fax/apps/app_confbridge.c
team/file/gulp_fax/apps/app_dial.c
team/file/gulp_fax/apps/app_directed_pickup.c
team/file/gulp_fax/apps/app_meetme.c
team/file/gulp_fax/apps/app_queue.c
team/file/gulp_fax/apps/app_voicemail.c
team/file/gulp_fax/apps/confbridge/conf_chan_announce.c
[... 15598 lines stripped ...]
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