[svn-commits] may: branch may/ooh323_qsig r369542 - in /team/may/ooh323_qsig: ./ apps/ chan...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Jul 2 17:38:53 CDT 2012
Author: may
Date: Mon Jul 2 17:38:13 2012
New Revision: 369542
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=369542
Log:
Multiple revisions 368181,368221,368268-368269,368311,368359,368421,368435,368441,368455,368466-368467,368472,368500,368519,368529,368537,368550,368566,368569,368588,368606,368637,368646,368663,368668,368673-368675,368680-368681,368688,368712,368714,368722,368751,368772,368784,368793-368794,368809,368832,368854-368855,368886,368896,368900,368920-368921,368929,368948,368966,368972,368979,368985,368991,369000,369007
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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 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 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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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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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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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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r368588 | kmoore | 2012-06-06 20:11:01 +0400 (Wed, 06 Jun 2012) | 15 lines
Ensure overlapping hold flags do not conflict
When changing between different modes of hold, the flags were not being
cleared out properly causing a failure to change hold states.
(closes issue ASTERISK-19919)
Patch-by: Morten Tryfoss
Reported-by: Morten Tryfoss
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r368606 | mjordan | 2012-06-06 21:22:11 +0400 (Wed, 06 Jun 2012) | 21 lines
Add feature modifier to versions produced from branches
Certain branches, such as Certified Asterisk, may have a modifier added to
them that specifies the features available in that branch. For branches, this
modifier is expected to be reflected in the location of the branch in
subversion. For example, a subversion of URL of /certified/branches/1.8.11
would have a feature modifier of 'certified'. This is slightly different then
how features are determined for tags, where the feature is part of the actual
tag name, e.g., "10.5.0-digiumphones".
In keeping with the nomenclature used for tags, the feature specifier for
branches is translated and placed after the revision numbers. For the example
given previously, this would result in a branch version of
"Asterisk SVN-branch-1.8.11-cert-rXXXXXX".
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r368637 | mmichelson | 2012-06-06 23:25:44 +0400 (Wed, 06 Jun 2012) | 34 lines
Fix a specific scenario where ACKs are not matched.
If a dialog-starting INVITE contains a to-tag, then Asterisk
will respond with a 481. In this case, the resulting incoming
ACK would not be matched, so Asterisk would continue retransmitting
the 481 until the transaction times out.
There were two issues. Asterisk, upon creating a sip_pvt would generate
a local tag. However, when the time came to transmit the 481, since there
was a to-tag in the INVITE, Asterisk would place this original to-tag
in the 481 response. When the ACK came in, Asterisk would attempt to
match the to-tag in the ACK to the generated local tag. Unfortunately,
Asterisk never actually transmitted a response with the generated local
tag, so the to-tag in the ACK would not match.
The other problem was that when the 481 was sent, nothing was set
on the sip_pvt to indicate what CSeq is expected in the ACK.
To fix the first problem, we zero out the to-tag seen in the incoming
INVITE. This way, Asterisk, when time to send a response, will send
its generated local tag instead.
To fix the second problem, we set the sip_pvt's pendinginvite to the
CSeq of the INVITE when we send a 481.
(closes issue ASTERISK-19892)
Reported by Mark Michelson
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r368646 | rmudgett | 2012-06-07 01:34:10 +0400 (Thu, 07 Jun 2012) | 20 lines
Fix POTS flash hook to orignate a second call deadlock.
A deadlock can occur when a POTS phone tries to flash hook to originate a
second call for 3-way or transfer. If another process is scanning the
channels container when the POTS line flash hooks then a deadlock will
occur.
* Release the channel and private locks when creating a new channel as a
result of a flash hook.
(closes issue ASTERISK-19842)
Reported by: rmudgett
Tested by: rmudgett
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r368663 | twilson | 2012-06-07 19:43:37 +0400 (Thu, 07 Jun 2012) | 9 lines
Add default handler documentation and standardize acl handler
Added documentation describing what flags and arguments to pass to
aco_option_register for default option types. Also changed the ACL
handler to use the flags parameter to differentiate between "permit"
and "deny" instead of adding an additional vararg parameter.
Review: https://reviewboard.asterisk.org/r/1969/
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r368668 | tzafrir | 2012-06-08 00:00:29 +0400 (Fri, 08 Jun 2012) | 4 lines
Fix a typo in format_ogg_vorbis.c: suport
Review: https://reviewboard.asterisk.org/r/1970/
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r368673 | twilson | 2012-06-08 00:32:07 +0400 (Fri, 08 Jun 2012) | 8 lines
Fix reloading an unchanged file with the Config Options API
Adding multiple file support broke reloading an unchanged file. This
adds an enum for return values for the aco_process_* functions and
ensures that the config is not applied if res is not ACO_PROCESS_OK.
Review: https://reviewboard.asterisk.org/r/1979/
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r368674 | rmudgett | 2012-06-08 00:37:05 +0400 (Fri, 08 Jun 2012) | 6 lines
Fix inverted test in app_queue for ringinuse.
Regression from -r367080 ringinuse commit.
(issue ASTERISK-19536)
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r368675 | rmudgett | 2012-06-08 00:39:25 +0400 (Fri, 08 Jun 2012) | 2 lines
Fix app_queue debug message use of args.options after the string has been parsed.
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r368680 | wedhorn | 2012-06-08 01:23:42 +0400 (Fri, 08 Jun 2012) | 10 lines
Skinny cleanup.
Removed d->registered which was mirroring d->session. Changed relevant
references to use d->session instead.
Moved setting and unsetting of l->device from session register to device
configuration. As such, l->device will always be valid unless it is has not
been configured to a device. Revised various test where checking if a device
is registered to use l->device->session.
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r368681 | wedhorn | 2012-06-08 01:44:15 +0400 (Fri, 08 Jun 2012) | 7 lines
Skinny cleanup (mwi_event_cb).
Original was testing for d->session, setting and testing again (all nested).
Removed duplicate testing and restructured function to test/return and then
the main code.
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r368688 | igorg | 2012-06-08 12:32:49 +0400 (Fri, 08 Jun 2012) | 8 lines
Fix MWI update so LED display correct voicemail state after phone usage. Also fixes few warnings.
(closes issue #19675)
Reported by: dbohling
Patches:
fixmwi.patch uploaded by dbohling (license 6378)
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r368712 | rmudgett | 2012-06-09 00:49:00 +0400 (Sat, 09 Jun 2012) | 1 line
Tweak ast_channel_softhangup_withcause_locked() to take a typed parameter.
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r368714 | rmudgett | 2012-06-09 01:08:17 +0400 (Sat, 09 Jun 2012) | 7 lines
Fix error paths in action_hangup() for AMI Hangup action.
* Check allocation function return values for failure. Crashing is bad.
* Tweak ast_regex_string_to_regex_pattern() parameters for proper ast_str
usage.
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r368722 | kmoore | 2012-06-11 18:12:08 +0400 (Mon, 11 Jun 2012) | 11 lines
Recorded merge of revisions 368721 from http://svn.asterisk.org/svn/asterisk/branches/10
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Fix compilation in dev-mode
Backport a compilation fix in md5.c from trunk that only showed up in
dev-mode under certain compiler versions.
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Merged revisions 368719 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r368751 | kmoore | 2012-06-11 19:23:30 +0400 (Mon, 11 Jun 2012) | 13 lines
Fix coverity UNUSED_VALUE findings in core support level files
Most of these were just saving returned values without using them and
in some cases the variable being saved to could be removed as well.
(issue ASTERISK-19672)
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r368772 | rmudgett | 2012-06-11 21:34:08 +0400 (Mon, 11 Jun 2012) | 20 lines
Fix deadlock potential with ast_set_hangupsource() calls.
Calling ast_set_hangupsource() with the channel lock held can result in a
deadlock because the function also locks the bridged channel.
(issue ASTERISK-19537)
(closes issue AST-891)
Reported by: Guenther Kelleter
Tested by: Guenther Kelleter
(closes issue ASTERISK-19801)
Reported by: Alec Davis
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r368784 | kmoore | 2012-06-12 08:03:23 +0400 (Tue, 12 Jun 2012) | 10 lines
Parse ANI2 information from SIP From header parameters
ANI2 information is now parsed out of SIP From headers when present in
the oli, isup-oli, and ss7-oli parameters and is available via the
CALLERID(ani2) dialplan function.
(closes issue ASTERISK-19912)
Patch-by: Rob Gagnon
Review: https://reviewboard.asterisk.org/r/1947/
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r368793 | mjordan | 2012-06-12 18:07:13 +0400 (Tue, 12 Jun 2012) | 18 lines
Fix deadlock in SIP transfers that involve a REFER request
In r367163, "send to voicemail" functionality was added to the SIP channel
driver. This required updating the party redirecting information for the
channel based on the headers provided in the REFER request. When the
redirecting party information is updated on the channel, a call to
ast_indicate_data occurs. Because handle_request_refer still had the sip_pvt
locked, a deadlock could occur between the pbx_thread and the do_monitor thread
servicing the REFER request.
This patch preserves the proper locking order between the channel and the
sip_pvt by ensuring that the sip_pvt is unlocked prior to updating the party
redirecting information on the channel.
(closes issue AST-903)
Reported by: Matt Jordan
patches:
jira_ast_903_trunk.patch by rmudgett (license 5621)
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r368794 | mjordan | 2012-06-12 18:09:41 +0400 (Tue, 12 Jun 2012) | 1 line
Update merge property information
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r368809 | mmichelson | 2012-06-12 19:46:48 +0400 (Tue, 12 Jun 2012) | 18 lines
Set the Caller ID "tag" on peers even if remote party information is present.
On incoming calls, we were setting the cid_tag on the dialog only if there was
no remote party information (Remote-Party-ID or P-Asserted-Identity) present.
The Caller ID tag is an invented parameter, though, and should be set no matter
the circumstance.
(closes issue ASTERISK-19859)
Reported by Thomas Arimont
(closes issue AST-884)
Reported by Trey Blancher
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r368832 | mjordan | 2012-06-12 22:41:50 +0400 (Tue, 12 Jun 2012) | 27 lines
Do not perform install on existing directories
If a directory already exists, performing a 'make install' will remove the
permissions associated with the current directory and replace them with the
permissions of the user executing the install.
This patch changes this behavior to only perform an install on the directory
if the directory does not exist. Thus, if a user later changes the permissions
on that directory, those permissions will be preserved in subsequent installs.
Review: https://reviewboard.asterisk.org/r/1986
Review: https://reviewboard.asterisk.org/r/1864
(closes issue ASTERISK-19492)
Reported by: Karl Fife
Tested by: Paul Belanger, Tilghman Lesher
patches:
ASTERISK-19492 by pabelanger
(uploaded by mjordan)
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r368854 | mjordan | 2012-06-13 18:31:24 +0400 (Wed, 13 Jun 2012) | 14 lines
Do not install empty directories; add ASTLIBDIR
r368830 modified the installation script to only create a directory if that
directory does not exist. If some directory variable was empty, it would attempt
to create the empty location. It also failed to create the ASTLIBDIR directory.
This patch fixes it such that the correct directories are made and only created if
a value specifying them actually exists.
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r368855 | mjordan | 2012-06-13 18:55:30 +0400 (Wed, 13 Jun 2012) | 4 lines
Replace MODULES_DIR with ASTMODDIR in Makefile's INSTALLDIRS
Post Asterisk 10, the MODULES_DIR variable no longer exists, and was replaced
with ASTMODDIR.
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r368886 | mmichelson | 2012-06-13 23:51:08 +0400 (Wed, 13 Jun 2012) | 14 lines
Remove forced linking of res_adsi.o
In GCC 4.5+ the result is that Asterisk has a phantom
module loaded at startup, claiming to be res_adsi.
(closes issue ASTERISK-19920)
reported by Leif Madsen
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r368896 | mjordan | 2012-06-14 00:28:07 +0400 (Thu, 14 Jun 2012) | 24 lines
Mark res_smdi/res_adsi as 'core' supported modules
Recently, various issues surrounding weak symbols have caused problems with
modules that rely on that feature to be enabled in menuselect. This includes
app_voicemail and chan_dahdi, as they both rely upon res_smdi and res_adsi,
which, in certain circumstances, may not be enabled by default in menuselect.
Because res_smdi/res_adsi are dependencies for chan_dahdi/app_voicemail, this
patch marks both as 'core' supported modules. This will allow both
app_voicemail and chan_dahdi to be enabled as well, regardless of whether or
not that system supports weak symbols.
(issue AST-900)
Reported by: Thomas Arimont
(issue AST-885)
Reported by: Denis Alberto Martinez
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r368900 | mmichelson | 2012-06-14 01:17:13 +0400 (Thu, 14 Jun 2012) | 22 lines
Fix a deadlock that occurs when func_volume is used on a local channel.
This was discovered by trying to perform a call forward to an extension
that makes use of func_volume. When the local channel is optimized away,
the datastore on the local;2 channel would have its audiohook destroyed
rather than detaching the audiohook from the channel and then destroying
it.
With this patch, func_volume's datastore destructor takes the proper
route of detaching the audiohook and then destroying it.
(closes issue ASTERISK-19611)
reported by Volker Sauer
Patches:
ASTERISK-19611.patch uploaded by Mark Michelson (license #5049)
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r368920 | twilson | 2012-06-14 17:35:07 +0400 (Thu, 14 Jun 2012) | 8 lines
Add filename alias support to the Config Options API
This adds the ability to handle a single filename alias for a config
file. This is useful if a config filename has changed, but the old
filename should be supported for backwards compatibility.
Review: https://reviewboard.asterisk.org/r/1981/
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r368921 | twilson | 2012-06-14 17:41:47 +0400 (Thu, 14 Jun 2012) | 7 lines
Add a post_apply callback to the Config Options API
This adds a callback that only fires when changes have been successfully
applied via the Config Options API.
Review: https://reviewboard.asterisk.org/r/1980/
........
r368929 | mmichelson | 2012-06-14 19:28:02 +0400 (Thu, 14 Jun 2012) | 13 lines
Revert Makefile change to remove embedding res_adsi.so
The change has resulted in a linking error for certain versions
of GCC. This is much worse than the original issue, so for now,
temporarily revert the change. A more thorough change will be
sought out.
........
Merged revisions 368927 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 368928 from http://svn.asterisk.org/svn/asterisk/branches/10
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r368948 | mjordan | 2012-06-14 21:34:10 +0400 (Thu, 14 Jun 2012) | 22 lines
AST-2012-009: Fix crash in chan_skinny due to Key Pad Button Message handling
AST-2012-008 (r367844) fixed a denial of service attack exploitable in the
Skinny channel driver that occurred when certain messages are sent after a
previously registered station sends an Off Hook message. Unresolved in that
patch is an issue in the Asterisk 10 releases, wherein, if a Station Key
Pad Button Message is processed after an Off Hook message, the channel driver
will inappropriately dereference a NULL pointer.
This patch fixes those places where the message handling or the channel
callback functions would attempt to dereference the line's pointer to the
device.
(issue ASTERISK-19905)
Reported by: Christoph Hebeisen
Tested by: mjordan, Christoph Hebeisen
Patches:
AST-2012-009-10.diff uploaded by mjordan (license 6283)
........
Merged revisions 368947 from http://svn.asterisk.org/svn/asterisk/branches/10
........
r368966 | qwell | 2012-06-14 23:40:11 +0400 (Thu, 14 Jun 2012) | 35 lines
Multiple revisions 368963,368965
........
r368963 | qwell | 2012-06-14 13:47:03 -0500 (Thu, 14 Jun 2012) | 14 lines
Remove global symbol requirement from app_voicemail.
This uses the existing "function installation" stuff that already existed for
other functions, like getting message counts.
(closes issue AST-807)
(issue AST-901)
(issue AST-908)
Review: https://reviewboard.asterisk.org/r/1965/
........
Merged revisions 368962 from http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
........
r368965 | qwell | 2012-06-14 14:04:57 -0500 (Thu, 14 Jun 2012) | 11 lines
These functions that were moved need to be static.
Also wrap test functions in a #ifdef.
(issue AST-807)
(issue AST-901)
(issue AST-908)
........
Merged revisions 368964 from http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
........
Merged revisions 368963,368965 from http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones
........
r368972 | rmudgett | 2012-06-15 00:49:28 +0400 (Fri, 15 Jun 2012) | 1 line
Move vm defines to group them better.
........
r368979 | rmudgett | 2012-06-15 02:57:21 +0400 (Fri, 15 Jun 2012) | 10 lines
Make the Hangup application set a softhangup flag.
The Hangup application used to just return -1 to cause normal dialplan
execution to hangup a channel. For the non-normal execution routines like
predial and connected-line interception routines, the hangup request would
exit the routine early but otherwise be ignored.
* Made the Hangup application not allow setting a cause code of zero. A
zero cause code is not defined.
........
r368985 | rmudgett | 2012-06-15 03:22:53 +0400 (Fri, 15 Jun 2012) | 27 lines
Allow non-normal execution routines to be able to run on hungup channels.
* Make non-normal dialplan execution routines be able to run on a hung up
channel. This is preparation work for hangup handler routines.
* Fixed ability to support relative non-normal dialplan execution
routines. (i.e., The context and exten are optional for the specified
dialplan location.) Predial routines are the only non-normal routines that
it makes sense to optionally omit the context and exten. Setting a hangup
handler also needs this ability.
* Fix Return application being able to restore a dialplan location
exactly. Channels without a PBX may not have context or exten set.
* Fixes non-normal execution routines like connected line interception and
predial leaving the dialplan execution stack unbalanced. Errors like
missing Return statements, popping too many stack frames using StackPop,
or an application returning non-zero could leave the dialplan stack
unbalanced.
* Fixed the AGI gosub application so it cleans up the dialplan execution
stack and handles the autoloop priority increments correctly.
* Eliminated the need for the gosub_virtual_context return location.
Review: https://reviewboard.asterisk.org/r/1984/
........
r368991 | rmudgett | 2012-06-15 04:55:43 +0400 (Fri, 15 Jun 2012) | 1 line
Remove remaining properties mmichelson left laying around from phones branch merge.
........
r369000 | qwell | 2012-06-15 19:33:41 +0400 (Fri, 15 Jun 2012) | 12 lines
Remove some symbol exports that got missed in the removal of global symbols.
(issue AST-807)
(issue AST-901)
(issue AST-908)
........
Merged revisions 368998 from http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
........
Merged revisions 368999 from http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones
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r369007 | kmoore | 2012-06-15 20:17:12 +0400 (Fri, 15 Jun 2012) | 11 lines
Add HANGUPCAUSE hash support to IAX2
Continuing with the Who Hung Up? project for Asterisk 11, this adds
support to IAX2 for the HANGUPCAUSE hash.
Additionally, this breaks out some functionality in frame.c for getting
information about frame types and subclasses.
Review: https://reviewboard.asterisk.org/r/1941/
(issue SWP-4222)
........
Merged revisions 368181,368221,368268-368269,368311,368359,368421,368435,368441,368455,368466-368467,368472,368500,368519,368529,368537,368550,368566,368569,368588,368606,368637,368646,368663,368668,368673-368675,368680-368681,368688,368712,368714,368722,368751,368772,368784,368793-368794,368809,368832,368854-368855,368886,368896,368900,368920-368921,368929,368948,368966,368972,368979,368985,368991,369000,369007 from http://svn.asterisk.org/svn/asterisk/trunk
Modified:
team/may/ooh323_qsig/ (props changed)
team/may/ooh323_qsig/apps/app_skel.c
team/may/ooh323_qsig/channels/chan_iax2.c
team/may/ooh323_qsig/include/asterisk/frame.h
team/may/ooh323_qsig/main/frame.c
team/may/ooh323_qsig/main/udptl.c
team/may/ooh323_qsig/tests/test_config.c
Propchange: team/may/ooh323_qsig/
------------------------------------------------------------------------------
--- branch-10-digiumphones-merged (original)
+++ branch-10-digiumphones-merged Mon Jul 2 17:38:13 2012
@@ -1,1 +1,1 @@
-/branches/10-digiumphones:364766,365396,368791,368963-368965,368999
+/branches/10-digiumphones:364766,365396,368791,368963-368965
Propchange: team/may/ooh323_qsig/
------------------------------------------------------------------------------
--- svn:mergeinfo (original)
+++ svn:mergeinfo Mon Jul 2 17:38:13 2012
@@ -1,2 +1,2 @@
/team/mmichelson/private/phones-trunk:358764-361321
-/trunk:357542,368181,368421,368435-368529,368537-368550,368991-369000
+/trunk:357542,368181-369007
Propchange: team/may/ooh323_qsig/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Mon Jul 2 17:38:13 2012
@@ -1,1 +1,1 @@
-/trunk:1-368158,368181,368991-369000
+/trunk:1-368158,368181-369007
Modified: team/may/ooh323_qsig/apps/app_skel.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/apps/app_skel.c?view=diff&rev=369542&r1=369541&r2=369542
==============================================================================
--- team/may/ooh323_qsig/apps/app_skel.c (original)
+++ team/may/ooh323_qsig/apps/app_skel.c Mon Jul 2 17:38:13 2012
@@ -633,7 +633,7 @@
static int reload_module(void)
{
- if (aco_process_config(&cfg_info, 1)) {
+ if (aco_process_config(&cfg_info, 1) == ACO_PROCESS_ERROR) {
return AST_MODULE_LOAD_DECLINE;
}
@@ -673,7 +673,7 @@
aco_option_register(&cfg_info, "max_number", ACO_EXACT, level_options, NULL, OPT_UINT_T, 0, FLDSET(struct skel_level, max_num));
aco_option_register(&cfg_info, "max_guesses", ACO_EXACT, level_options, NULL, OPT_UINT_T, 1, FLDSET(struct skel_level, max_guesses));
- if (aco_process_config(&cfg_info, 0)) {
+ if (aco_process_config(&cfg_info, 0) == ACO_PROCESS_ERROR) {
goto error;
}
Modified: team/may/ooh323_qsig/channels/chan_iax2.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/channels/chan_iax2.c?view=diff&rev=369542&r1=369541&r2=369542
==============================================================================
--- team/may/ooh323_qsig/channels/chan_iax2.c (original)
+++ team/may/ooh323_qsig/channels/chan_iax2.c Mon Jul 2 17:38:13 2012
@@ -10135,6 +10135,62 @@
}
#endif
+ if (iaxs[fr->callno]->owner && (fh->type == AST_FRAME_IAX || fh->type == AST_FRAME_CONTROL)) {
+ struct ast_control_pvt_cause_code *cause_code;
+ int data_size = sizeof(*cause_code);
+ char subclass[40] = "";
+
+ /* get subclass text */
+ if (fh->type == AST_FRAME_IAX) {
+ iax_frame_subclass2str(fh->csub, subclass, sizeof(subclass));
+ } else {
+ struct ast_frame tmp_frame = {0,};
+ tmp_frame.frametype = fh->type;
+ tmp_frame.subclass.integer = fh->csub;
+ ast_frame_subclass2str(&tmp_frame, subclass, sizeof(subclass), NULL, 0);
+ }
+
+ /* add length of "IAX2 " */
+ data_size += 5;
+ if (fh->type == AST_FRAME_CONTROL) {
+ /* add length of "Control " */
+ data_size += 8;
+ } else if (fh->csub == IAX_COMMAND_HANGUP
+ || fh->csub == IAX_COMMAND_REJECT
+ || fh->csub == IAX_COMMAND_REGREJ
+ || fh->csub == IAX_COMMAND_TXREJ) {
+ /* for IAX hangup frames, add length of () and number */
+ data_size += 3;
+ if (ies.causecode > 9) {
+ data_size++;
+ }
+ if (ies.causecode > 99) {
+ data_size++;
+ }
+ }
+ /* add length of subclass */
+ data_size += strlen(subclass);
+
+ cause_code = alloca(data_size);
+ ast_copy_string(cause_code->chan_name, ast_channel_name(iaxs[fr->callno]->owner), AST_CHANNEL_NAME);
+
+ if (fh->type == AST_FRAME_IAX &&
+ (fh->csub == IAX_COMMAND_HANGUP
+ || fh->csub == IAX_COMMAND_REJECT
+ || fh->csub == IAX_COMMAND_REGREJ
+ || fh->csub == IAX_COMMAND_TXREJ)) {
+ snprintf(cause_code->code, data_size - sizeof(*cause_code) + 1, "IAX2 %s(%d)", subclass, ies.causecode);
+ } else {
+ snprintf(cause_code->code, data_size - sizeof(*cause_code) + 1, "IAX2 %s%s", (fh->type == AST_FRAME_CONTROL ? "Control " : ""), subclass);
+ }
+
+ iax2_queue_control_data(fr->callno, AST_CONTROL_PVT_CAUSE_CODE, cause_code, data_size);
+ if (!iaxs[fr->callno]) {
+ ast_variables_destroy(ies.vars);
+ ast_mutex_unlock(&iaxsl[fr->callno]);
+ return 1;
+ }
+ }
/* count this frame */
iaxs[fr->callno]->frames_received++;
Modified: team/may/ooh323_qsig/include/asterisk/frame.h
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/include/asterisk/frame.h?view=diff&rev=369542&r1=369541&r2=369542
==============================================================================
--- team/may/ooh323_qsig/include/asterisk/frame.h (original)
+++ team/may/ooh323_qsig/include/asterisk/frame.h Mon Jul 2 17:38:13 2012
@@ -597,9 +597,31 @@
int ast_frame_slinear_sum(struct ast_frame *f1, struct ast_frame *f2);
/*!
- * \brief Clear all audio samples from an ast_frame. The frame must be AST_FRAME_VOICE and AST_FORMAT_SLINEAR
+ * \brief Clear all audio samples from an ast_frame. The frame must be AST_FRAME_VOICE and AST_FORMAT_SLINEAR
*/
int ast_frame_clear(struct ast_frame *frame);
+
+/*!
+ * \brief Copy the discription of a frame's subclass into the provided string
+ *
+ * \param f The frame to get the information from
+ * \param subclass Buffer to fill with subclass information
+ * \param slen Length of subclass buffer
+ * \param moreinfo Buffer to fill with additional information
+ * \param mlen Length of moreinfo buffer
+ * \since 11
+ */
+void ast_frame_subclass2str(struct ast_frame *f, char *subclass, size_t slen, char *moreinfo, size_t mlen);
+
+/*!
+ * \brief Copy the discription of a frame type into the provided string
+ *
+ * \param frame_type The frame type to be described
+ * \param ftype Buffer to fill with frame type description
+ * \param len Length of subclass buffer
+ * \since 11
+ */
+void ast_frame_type2str(enum ast_frame_type frame_type, char *ftype, size_t len);
#if defined(__cplusplus) || defined(c_plusplus)
}
Modified: team/may/ooh323_qsig/main/frame.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/main/frame.c?view=diff&rev=369542&r1=369541&r2=369542
==============================================================================
--- team/may/ooh323_qsig/main/frame.c (original)
+++ team/may/ooh323_qsig/main/frame.c Mon Jul 2 17:38:13 2012
@@ -522,96 +522,70 @@
dst_s[i] = (src_s[i]<<8) | (src_s[i]>>8);
}
-/*! Dump a frame for debugging purposes */
-void ast_frame_dump(const char *name, struct ast_frame *f, char *prefix)
-{
- const char noname[] = "unknown";
- char ftype[40] = "Unknown Frametype";
- char cft[80];
- char subclass[40] = "Unknown Subclass";
- char csub[80];
- char moreinfo[40] = "";
- char cn[60];
- char cp[40];
[... 361 lines stripped ...]
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