[svn-commits] may: branch may/ooh323_ipv6_direct_rtp r369553 - /team/may/ooh323_ipv6_direct...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jul 3 07:08:31 CDT 2012
Author: may
Date: Tue Jul 3 07:08:20 2012
New Revision: 369553
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=369553
Log:
Multiple revisions 368181,368221,368268-368269,368311,368359,368421,368435,368441,368455,368466-368467,368472,368500,368519,368529,368537,368550,368566,368569
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r368181 | twilson | 2012-06-01 20:33:25 +0400 (Fri, 01 Jun 2012) | 8 lines
Add new config-parsing framework
This framework adds a way to register the various options in a config
file with Asterisk and to handle loading and reloading of that config
in a consistent and atomic manner.
Review: https://reviewboard.asterisk.org/r/1873/
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r368221 | twilson | 2012-06-01 22:20:44 +0400 (Fri, 01 Jun 2012) | 2 lines
Add missing config for config API test
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r368268 | kpfleming | 2012-06-02 00:31:15 +0400 (Sat, 02 Jun 2012) | 23 lines
Improve SDP parsing warning messages
* 'Unsupported media type' is only reported when that is in fact the case,
not when a supported media type is included in an 'm' line that has an
invalid format.
* All warning messages related to parsing 'm' lines now include the 'm' line contents.
* (minor bugfix) newline added to port-number-zero warning messages.
* Warning messages improved to use RFC-specified terminology for various items.
* Warnings for offers that include more than one port for a single media type now
include the media type.
Review: https://reviewboard.asterisk.org/r/1811/
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Merged revisions 368218 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 368267 from http://svn.asterisk.org/svn/asterisk/branches/10
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r368269 | kpfleming | 2012-06-02 00:42:10 +0400 (Sat, 02 Jun 2012) | 10 lines
Improve SDP offer/answer RFC compliance
Asterisk should not accept SDP offers that contain unknown RTP profiles (for
audio/video streams) or unknown top-level media types. When it does, it answers
with an SDP that does not match the offer properly, and this will nearly
always result in a broken call. This patch causes such offers to be rejected.
Review: https://reviewboard.asterisk.org/r/1811/
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r368311 | rmudgett | 2012-06-02 03:53:59 +0400 (Sat, 02 Jun 2012) | 18 lines
Fix deadlock when Gosub used with alternate dialplan switches.
Attempting to remove a channel from autoservice with the channel lock held
will result in deadlock.
* Restructured gosub_exec() to not call ast_parseable_goto() and
ast_exists_extension() with the channel lock held.
(closes issue ASTERISK-19764)
Reported by: rmudgett
Tested by: rmudgett
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Merged revisions 368308 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 368310 from http://svn.asterisk.org/svn/asterisk/branches/10
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r368359 | file | 2012-06-03 01:13:36 +0400 (Sun, 03 Jun 2012) | 4 lines
Add res_http_websocket module which implements the WebSocket protocol according to RFC 6455.
Review: https://reviewboard.asterisk.org/r/1952/
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r368421 | rmudgett | 2012-06-04 23:46:33 +0400 (Mon, 04 Jun 2012) | 26 lines
Fix potential deadlock between masquerade and chan_local.
* Restructure ast_do_masquerade() to not hold channel locks while it calls
ast_indicate().
* Simplify many calls to ast_do_masquerade() since it will never return a
failure now. If it does fail internally because a channel driver callback
operation failed, the only thing ast_do_masquerade() can do is generate a
warning message about strange things may happen and press on.
* Fixed the call to ast_bridged_channel() in ast_do_masquerade(). This
change fixes half of the deadlock reported in ASTERISK-19801 between
masquerades and chan_iax.
(closes issue ASTERISK-19537)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1915/
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Merged revisions 368405 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 368407 from http://svn.asterisk.org/svn/asterisk/branches/10
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r368435 | mmichelson | 2012-06-05 00:26:12 +0400 (Tue, 05 Jun 2012) | 35 lines
Merge changes dealing with support for Digium phones.
Presence support has been added. This is accomplished by
allowing for presence hints in addition to device state
hints. A dialplan function called PRESENCE_STATE has been
added to allow for setting and reading presence. Presence
can be transmitted to Digium phones using custom XML
elements in a PIDF presence document.
Voicemail has new APIs that allow for moving, removing,
forwarding, and playing messages. Messages have had a new
unique message ID added to them so that the APIs will work
reliably. The state of a voicemail mailbox can be obtained
using an API that allows one to get a snapshot of the mailbox.
A voicemail Dialplan App called VoiceMailPlayMsg has been
added to be able to play back a specific message.
Configuration hooks have been added. Configuration hooks
allow for a piece of code to be executed when a specific
configuration file is loaded by a specific module. This is
useful for modules that are dependent on the configuration
of other modules.
chan_sip now has a public method that allows for a custom
SIP INFO request to be sent mid-dialog. Digium phones use
this in order to display progress bars when files are played.
Messaging support has been expanded a bit. The main
visible difference is the addition of an AMI action
MessageSend.
Finally, a ParkingLots manager action has been added in order
to get a list of parking lots.
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r368441 | mmichelson | 2012-06-05 00:30:07 +0400 (Tue, 05 Jun 2012) | 3 lines
Remove automerge properties.
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r368455 | mmichelson | 2012-06-05 00:40:12 +0400 (Tue, 05 Jun 2012) | 3 lines
Remove some extra debugging I forgot to remove in the merge of Digium phone support.
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r368466 | mmichelson | 2012-06-05 00:51:17 +0400 (Tue, 05 Jun 2012) | 8 lines
Add vim syntax highlighting for type=line, type=phone, and type=application.
(closes issue ASTERISK-19800)
Reported by: Billy Chia
Patches:
asterisk.vim.patch uploaded by Billy Chia (license #6381)
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r368467 | mmichelson | 2012-06-05 00:53:43 +0400 (Tue, 05 Jun 2012) | 3 lines
Also have vim syntax-highlight type=network.
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r368472 | rmudgett | 2012-06-05 01:18:04 +0400 (Tue, 05 Jun 2012) | 13 lines
Document BLINDTRANSFER behavior change.
(issue ASTERISK-19322)
(closes issue ASTERISK-19875)
Reported by: call
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Merged revisions 368469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 368470 from http://svn.asterisk.org/svn/asterisk/branches/10
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r368500 | mmichelson | 2012-06-05 02:12:19 +0400 (Tue, 05 Jun 2012) | 19 lines
Relay proper SIP responses on calling side.
Revision 351130 broke corect HANGUPCAUSE setting
for the 404 case in chan_sip. Other cases were also
potentially broken. This patch fixes the relaying
of causes to be what they used to be.
(closes issue ASTERISK-19914)
Reported by Pavel Troller
Tested by Walter Doekes (via a reviewboard test to be committed later)
Patches:
chan_sip.diff uploaded by Pavel Troller (license #6302)
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Merged revisions 368498 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 368499 from http://svn.asterisk.org/svn/asterisk/branches/10
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r368519 | kmoore | 2012-06-05 18:41:43 +0400 (Tue, 05 Jun 2012) | 11 lines
Convert AST_FLAG_ANSWERED_ELSEWHERE usage to AST_CAUSE_ANSWERED_ELSEWHERE
This was essentially duplicated functionality where normal channels used
AST_CAUSE_ANSWERED_ELSEWHERE while local channels and queues used
AST_FLAG_ANSWERED_ELSEWHERE. This removes the flag and converts that usage
into AST_CAUSE_ANSWERED_ELSEWHER usage.
Review: https://reviewboard.asterisk.org/r/1944
(closes issue ASTERISK-19865)
Patch-by: Birger Harzenetter
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r368529 | kmoore | 2012-06-05 19:23:43 +0400 (Tue, 05 Jun 2012) | 14 lines
Ensure that pages and emails are sent using RFC822-compliant date format
When localization was added to app_voicemail, these headers were altered
when they should have remained in en_US format for RFC compliance. This
reverts the changes to those two lines.
(closes issue ASTERISK-19876)
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Merged revisions 368520 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 368524 from http://svn.asterisk.org/svn/asterisk/branches/10
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r368537 | kmoore | 2012-06-05 19:28:28 +0400 (Tue, 05 Jun 2012) | 11 lines
Recorded merge of revisions 368536 from http://svn.asterisk.org/svn/asterisk/branches/10
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Resolve some build warnings
My newly upgraded compiler caught these usages of uninitialized values.
They weren't actually used.
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Merged revisions 368533 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r368550 | jrose | 2012-06-05 20:25:14 +0400 (Tue, 05 Jun 2012) | 3 lines
Merge 'core' and 'core changes' sections in CHANGES file.
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r368566 | rmudgett | 2012-06-06 04:54:20 +0400 (Wed, 06 Jun 2012) | 1 line
Make builtin_blindtransfer() fully use ast_async_goto() abilities.
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r368569 | rmudgett | 2012-06-06 05:11:12 +0400 (Wed, 06 Jun 2012) | 18 lines
Fix parked call performing a DTMF blind transfer after being retrieved.
When a parked call was retrieved from the parking lot, it could not do a
blind transfer because it caused the involved calls to be hung up
unconditionally.
* Made the ParkedCall application return the ast_bridge_call() return
value.
(closes issue ABE-2862)
Reported by: Vlad Povorozniuc
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Merged revisions 368567 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 368568 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 368181,368221,368268-368269,368311,368359,368421,368435,368441,368455,368466-368467,368472,368500,368519,368529,368537,368550,368566,368569 from http://svn.asterisk.org/svn/asterisk/trunk
Modified:
team/may/ooh323_ipv6_direct_rtp/ (props changed)
Propchange: team/may/ooh323_ipv6_direct_rtp/
------------------------------------------------------------------------------
--- svn:mergeinfo (original)
+++ svn:mergeinfo Tue Jul 3 07:08:20 2012
@@ -1,2 +1,2 @@
/team/mmichelson/private/phones-trunk:358764-361321
-/trunk:331201-331202,346391,354429,356042,357272,360190,362888,362919-362920,368421-368569,368588-369034
+/trunk:331201-331202,346391,354429,356042,357272,360190,362888,362919-362920,368181-368569,368588-369034
Propchange: team/may/ooh323_ipv6_direct_rtp/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Jul 3 07:08:20 2012
@@ -1,1 +1,1 @@
-/trunk:1-368158,368588-369034
+/trunk:1-368158,368181-368569,368588-369034
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