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r379312 | mmichelson | 2013-01-16 16:51:32 -0600 (Wed, 16 Jan 2013) | 11 lines
Further fix misinformation in the description of manager MailboxStatus command.
The description still claimed that it returned the number of messages rather than
whether there were messages waiting.
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r379278 | qwell | 2013-01-16 15:13:53 -0600 (Wed, 16 Jan 2013) | 11 lines
Reduce number of packages install_prereq installs on Debian systems.
'search' will look for any package containing the name provided, so we need to
force a more exact search.
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r379229 | mjordan | 2013-01-16 11:46:15 -0600 (Wed, 16 Jan 2013) | 10 lines
Let documentation reference links specify which module they're linking to
Again, since res_jabber/res_xmpp have duplicate APIs, their documentation ref
links have to specify which reference they're referring to. The various
documentation parsers can interpret the module attribute however they want
in order to construct the appropriate links.
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r379231 | rmudgett | 2013-01-16 11:49:52 -0600 (Wed, 16 Jan 2013) | 10 lines
chan_misdn: Fix compile error.
(issue ASTERISK-15456)
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r379233 | rmudgett | 2013-01-16 12:09:28 -0600 (Wed, 16 Jan 2013) | 8 lines
Reduce call-id logging resource usage.
Since there is no need for the call-id logging ao2 object to have a lock,
don't create it with one.
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r379211 | mjordan | 2013-01-16 09:33:05 -0600 (Wed, 16 Jan 2013) | 21 lines
Multiple revisions 379209-379210
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r379209 | mjordan | 2013-01-16 09:27:44 -0600 (Wed, 16 Jan 2013) | 8 lines
Add module tags to documentation for res_jabber/res_xmpp
Since res_jabber/res_xmpp provide the same APIs (app/func/manager/etc.),
the XML documentation for each needs to call out which module is providing
the documentation. The module attribute has been added to the various XML
fragments for this purpose.
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r379210 | mjordan | 2013-01-16 09:30:20 -0600 (Wed, 16 Jan 2013) | 4 lines
Update the dtd to actually *support* the module attribute in all elements
Mea culpa.
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r379180 | mjordan | 2013-01-15 22:14:38 -0600 (Tue, 15 Jan 2013) | 27 lines
Fix parsing SMSSRC for SMS messages
The parser for SMS messages would incorrectly parse out the from number.
The parsing would incorrectly start scanning for the from number at the
same index as the first double quote ("); this would inadvertently cause
it to treat the first double quote as the terminating double quote for
the from number as well.
The SMSSRC should now populate correctly.
(closes issue ASTERISK-16822)
Reported by: menschentier
Tested by: Jonas Falck
patches:
fixSMSSRC.patch uploaded by jonax (license 6320)
(closes issue ASTERISK-19153)
Reported by: Panos Gkikakis
patches:
sms-sender-fix.diff uploaded by roeften (license 5884)
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r379144 | mjordan | 2013-01-15 17:54:34 -0600 (Tue, 15 Jan 2013) | 17 lines
Add busy detection to chan_mobile
From the patch author:
"First this patch adds general support for busy detection. It also adds support
for the ECAM command at Sony Ericsson phones and also signals busy when only
early media was received but the call got not answered."
Review: https://reviewboard.asterisk.org/r/323
(closes issue ASTERISK-14527)
Reported by: Artem Makhutov
Tested by: Artem Makhutov
patches:
busy-full5.patch uploaded by artem (license 5757)
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r379147 | mjordan | 2013-01-15 18:16:22 -0600 (Tue, 15 Jan 2013) | 25 lines
Set the INVALID_EXTEN channel variable when chan_misdn forces the 'i' extension
The chan_misdn channel driver will send a channel with an invalid destination
to the 'i' extension itself if said extension can be reached. It forgot,
however, to set the INVALID_EXTEN channel variable when it bounces the channel
to this extension. Dialplan writers everywhere moaned at yet another
inconsistency.
This is yet another example of why duplicating logic in multiple places results
in bugs that stick around in Jira for just under three years.
Yes: ASTERISK-15456 was created on January 18th, 2010. Patch committed on
January 15th, 2013. Ouch.
(closes issue ASTERISK-15456)
Reported by: Thomas Omerzu
patches:
chan_misdn_invalid.patch2 uploaded by Thomas Omerzu (license 5927)
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r379128 | rmudgett | 2013-01-15 16:23:49 -0600 (Tue, 15 Jan 2013) | 1 line
Fix ast_bridge_features_register() not registering builtin features. I broke. Ooops.
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Mark Michelson [Tue, 15 Jan 2013 21:15:04 +0000 (21:15 +0000)]
Add doxygen to accessors and increase refcount of taskprocessor before returning.
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Mark Michelson [Tue, 15 Jan 2013 21:09:55 +0000 (21:09 +0000)]
Make the threadpool listener opaque.
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Mark Michelson [Tue, 15 Jan 2013 20:48:45 +0000 (20:48 +0000)]
Make ast_taskprocessor_listener opaque.
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Mark Michelson [Tue, 15 Jan 2013 20:15:00 +0000 (20:15 +0000)]
Address further review feedback from David Lee.
* Clarify some documentation
* Change copyright date of taskprocessor files
* Address potential issue of creating taskprocessor with listener if
taskprocessor with that name exists already
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Mark Michelson [Tue, 15 Jan 2013 19:44:25 +0000 (19:44 +0000)]
Make the initial size of the threadpool part of the options passed in.
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Mark Michelson [Tue, 15 Jan 2013 19:36:33 +0000 (19:36 +0000)]
Remove threadpool listener alloc and destroy callbacks.
This replaces the destroy callback with a shutdown callback
instead.
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Mark Michelson [Tue, 15 Jan 2013 18:40:36 +0000 (18:40 +0000)]
Remove alloc and destroy callbacks from the taskprocessor.
Now user data is allocated by the creator of the taskprocessor
listener and that user data is passed into ast_taskprocessor_listener_alloc().
Similarly, freeing of the user data is left up to the user himself. He can
free the data when the taskprocessor shuts down, or he can choose to hold
onto it if it makes sense to do so.
This, unsurprisingly, makes threadpool allocation a LOT cleaner now.
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r379070 | dlee | 2013-01-14 15:47:31 -0600 (Mon, 14 Jan 2013) | 1 line
Fixed doc comment for ast_test_validate
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r379021 | dlee | 2013-01-14 09:29:22 -0600 (Mon, 14 Jan 2013) | 15 lines
Fix XML encoding of 'identity display' in NOTIFY messages, continued.
When r378933 was merged into 1.8, it should have also escaped
remote_display, since it will have the same XML encoding problem when
the caller/callee roles are reversed.
(closes issue ABE-2902)
Reported by: Guenther Kelleter
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r379023 | dlee | 2013-01-14 09:58:01 -0600 (Mon, 14 Jan 2013) | 20 lines
Masquerades are an insane implementation detail within Asterisk. It generates
a number of useless and confusing events, and manipulates channels in a way
that semantically doesn't make sense. I've given a fairly thorough review of
masquerade code and its usage on the wiki at
https://wiki.asterisk.org/wiki/x/IwBRAQ.
While ultimately it makes the most sense to abandon masquerades altogether,
it will take some time to completely irradicate. Even then, there may always
be code that's not worth rewriting to get rid of the masquerade.
This patch does two things to make masquerades slightly less insane:
* When swapping the names of the original and clone channel, only emit a
single rename event of original -> original<ZOMBIE>. The original code
issued three rename events to accomplish the same end.
* In addition to swapping the names of the channels, also swap their
uniqueid's. This allows the 'Uniqueid' field to be used as a stable
identifier for a channel from and external interface, such as AMI.
Review: https://reviewboard.asterisk.org/r/2266/
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r378985 | mjordan | 2013-01-13 16:07:00 -0600 (Sun, 13 Jan 2013) | 20 lines
Reset RTP timestamp; sequence number on SSRC change
In r370252 for ASTERISK-18404, Asterisk's handling of RTP was modified to
better account for out of order RTP packets. This was accomplished by using the
RTP timestamp and sequence number to check for out of order packets. However,
when a SSRC change occurs, the timestamp and sequence number will no longer
have any relation to the previously received packets. The variables tracking
the timestamp and sequence number therefore have to be reset.
(closes issue ASTERISK-20906)
Reported by: Eelco Brolman
patches:
dtmf_on_hold.patch uploaded by Eelco Brolman (license #6442)
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r378935 | dlee | 2013-01-12 00:43:37 -0600 (Sat, 12 Jan 2013) | 41 lines
Fix XML encoding of 'identity display' in NOTIFY messages.
XML encoding in chan_sip is accomplished by naively building the XML
directly from strings. While this usually works, it fails to take into
account escaping the reserved characters in XML.
This patch adds an 'ast_xml_escape' function, which works similarly to
'ast_uri_encode'. This is used to properly escape the local_display
attribute in XML formatted NOTIFY messages.
Several things to note:
* The Right Thing(TM) to do would probably be to replace the
ast_build_string stuff with building an ast_xml_doc. That's a much
bigger change, and out of scope for the original ticket, so I
refrained myself.
* It is with great sadness that I wrote my own ast_xml_escape
function. There's one in libxml2, but it's knee-deep in
libxml2-ness, and not easily used to one-off escape a
string.
* I only escaped the string we know is causing problems
(local_display). At least some of the other strings are
URI-encoded, which should be XML safe. Rather than figuring out
what's safe and escaping what's not, it would be much cleaner to
simply build an ast_xml_doc for the messages and let the XML
library do the XML escaping. Like I said, that's out of scope.
(closes issue ABE-2902)
Reported by: Guenther Kelleter
Tested by: Guenther Kelleter
Review: http://reviewboard.digium.internal/r/365/
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r378915 | dlee | 2013-01-11 16:31:42 -0600 (Fri, 11 Jan 2013) | 21 lines
Add JSON API for Asterisk.
This provides a JSON API by pulling in and wrapping the Jansson JSON
library[1]. The Asterisk API basically mirrors the Jansson
functionality, with a few minor tweaks.
* Some names have been asteriskified to protect the innocent.
* Jansson provides both reference-stealing and reference-borrowing
versions of several API's. The Asterisk API is exclusively
reference-stealing for operations that put elements into arrays and
objects.
* No support for doubles, since we usually don't need that.
* Coming along for the ride is the ast_test_validate macro, which made
the unit tests much easier to write.
[1]: http://www.digip.org/jansson/
(issue ASTERISK-20887)
(closes issue ASTERISK-20888)
Review: https://reviewboard.asterisk.org/r/2264/
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r378918 | file | 2013-01-11 17:05:38 -0600 (Fri, 11 Jan 2013) | 11 lines
Retain XMPP filters across reconnections so external modules continue to function as expected.
Previously if an XMPP client reconnected any filters added by an external module were lost.
This issue exhibited itself with chan_motif not receiving and reacting to Jingle signaling.
(closes issue ASTERISK-20916)
Reported by: kuj
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r378889 | rmudgett | 2013-01-09 20:40:50 -0600 (Wed, 09 Jan 2013) | 8 lines
* Simplify native bridge code in ast_channel_bridge().
* Fix an unbalanced manager_bridge_event(unlink) call if
AST_SOFTHANGUP_UNBRIDGE is set in ast_channel_bridge().
* Make ast_channel_bridge() use common cleanup code when leaving the
bridge.
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r378874 | rmudgett | 2013-01-09 19:43:27 -0600 (Wed, 09 Jan 2013) | 4 lines
* Removed some noop code and restructured an else-if ladder in ast_generic_bridge().
* Trivial changes in ast_channel_bridge().
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r378854 | rmudgett | 2013-01-09 17:22:00 -0600 (Wed, 09 Jan 2013) | 1 line
Fix logger.c function definition.
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r378858 | rmudgett | 2013-01-09 17:23:41 -0600 (Wed, 09 Jan 2013) | 6 lines
Trivial misc bridge code changes.
* softmix_bridge_thread() was redundantly initializing an 8K buffer.
* Promoted a debug message to a warning in multiplexed_add_or_remove().
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r378859 | rmudgett | 2013-01-09 17:51:45 -0600 (Wed, 09 Jan 2013) | 6 lines
* Simple optimization of bridge_playfile().
* Squeezed some redundancy out of update_bridge_vars().
* Wrapped long line in __ast_change_name_nolink().
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r378840 | rmudgett | 2013-01-09 16:56:08 -0600 (Wed, 09 Jan 2013) | 2 lines
Trivial misc bridge code changes.
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r378823 | rmudgett | 2013-01-09 16:15:41 -0600 (Wed, 09 Jan 2013) | 2 lines
Tweaked __ast_test_suite_assert_notify() and __ast_test_suite_event_notify() to be void functions.
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r378783 | dlee | 2013-01-09 14:30:33 -0600 (Wed, 09 Jan 2013) | 14 lines
Fix end condition in ast_rtp_lookup_mime_multiple2.
The erroneous end condition would never include the AST_RTP_CISCO_DTMF flag
in the debug output.
(closes issue ASTERISK-20772)
Reported by: Xavier Hienne
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r378789 | rmudgett | 2013-01-09 14:56:23 -0600 (Wed, 09 Jan 2013) | 4 lines
* Found some more places to use ast_channel_lock_both().
* Minor optimization in ast_rtp_instance_early_bridge().
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r378790 | rmudgett | 2013-01-09 15:14:39 -0600 (Wed, 09 Jan 2013) | 4 lines
* Whitespace changes.
* Made ast_test_init() match its prototype.
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r378735 | dlee | 2013-01-09 13:38:53 -0600 (Wed, 09 Jan 2013) | 13 lines
Replace errant tabs with spaces in causes.h.
(closes issue ASTERISK-20826)
Reported by: snuffy
Patches:
notabs.dif uploaded by snuffy (license 5024)
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r378748 | dlee | 2013-01-09 14:12:00 -0600 (Wed, 09 Jan 2013) | 13 lines
Move declaration of ast_regex_string_to_regex_pattern futher down strings.h.
The prior location is before the declaration of struct ast_str, which causes
compiler warnings.
(closes issue ASTERISK-20852)
Reported by: Pavel Troller
Patches:
strings.diff uploaded by Pavel Troller (license 6302)
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r378688 | rmudgett | 2013-01-08 17:44:26 -0600 (Tue, 08 Jan 2013) | 35 lines
app_queue: Fix multiple calls to a queue member that is in only one queue.
When ringinuse=no queue members can receive more than one call if these
calls happen at nearly the same time.
* Fix so a queue member does not receive more than one call from a queue.
NOTE: This fix does not prevent multiple calls to a member if the member
is in more than one queue.
* Did some refactoring to eliminate some code redundancy.
(issue ASTERISK-16115)
Reported by: nik600
Patches:
jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett
Modified
* Revert the -r341580 and -r341599 changes adding the queues.conf
check_state_unknown option as it was added in an attempt to fix this
problem. The fix did not need to be optional. The fix should not have
tried to explicitly set the device state. Setting the device state by
something other than the device introduces a race condition. I also could
not see how the change would be effective other than delaying the
app_queue code long enough for the device state to propagate to app_queue.
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r378691 | rmudgett | 2013-01-08 18:05:35 -0600 (Tue, 08 Jan 2013) | 10 lines
app_queue: Fix incorrect assertion.
(issue ASTERISK-16115)
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Remove tasks from the taskprocessor and free them when taskprocessor is destroyed.
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Mark Michelson [Mon, 7 Jan 2013 22:56:37 +0000 (22:56 +0000)]
Add some doxygen and remove an unnecessary unlock.
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Mark Michelson [Mon, 7 Jan 2013 22:16:06 +0000 (22:16 +0000)]
Address review board feedback from Matt and Richard
* Remove extraneous whitespace
* Bump up debug levels of messages and add identifying info to messages.
* Account for potential failures of ao2_link()
* Add additional test and some more test data
* Add some comments in places where they could be useful
* Make threadpool listeners and their callbacks optional
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r378634 | wedhorn | 2013-01-06 15:37:59 -0600 (Sun, 06 Jan 2013) | 6 lines
Skinny blob cleanup
Cleanup of red blobs in chan_skinny and possible other small formatting issues.
Review: https://reviewboard.asterisk.org/r/2262/
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r378623 | wedhorn | 2013-01-06 14:45:12 -0600 (Sun, 06 Jan 2013) | 12 lines
Rewrite skinny dialing to remove threaded simpleswitch
This rewrite changes skinny dialing from the threaded simpleswitch
to a scheduled timeout approach. There were some underlying issues
with the threaded simple switch with occasional corruption and
possible segfaults.
Review: https://reviewboard.asterisk.org/r/2240/
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r378624 | wedhorn | 2013-01-06 15:09:43 -0600 (Sun, 06 Jan 2013) | 6 lines
Add group and namedgroup pickup to skinny
Above says it all. Code by snuff, cleaned up by me.
Review: https://reviewboard.asterisk.org/r/2246/
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r378593 | jrose | 2013-01-04 17:14:54 -0600 (Fri, 04 Jan 2013) | 23 lines
res_srtp: Prevent a crash from occurring due to srtp_create failures in srtp_create
Under some circumstances, libsrtp's srtp_create function deallocates memory that
it wasn't initially responsible for allocating. Because we weren't initially
aware of this behavior, this memory was still used in spite of being unallocated
during the course of the srtp_unprotect function. A while back I made a patch
which would set this value to NULL, but that exposed a possible condition where
we would then try to check a member of the struct which would cause a segfault.
In order to address these problems, ast_srtp_unprotect will now set an error value
when it ends without a valid SRTP session which will result in the caller of
srtp_unprotect observing this error and hanging up the relevant channel instead of
trying to keep using the invalid session address.
(closes issue ASTERISK-20499)
Reported by: Tootai
Review: https://reviewboard.asterisk.org/r/2228/diff/#index_header
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r378565 | elguero | 2013-01-04 15:20:12 -0600 (Fri, 04 Jan 2013) | 27 lines
Fix SIP Notify Messages To Have The Proper IP Address In The FROM Field
On a multihomed server when sending a NOTIFY message, we were not figuring out
which network should be used to contact the peer.
This patch fixes the problem by calling ast_sip_ouraddrfor() and then
build_via() so that our NOTIFY message contains the correct IP address.
Also, a debug message is being added to help follow the call-id changes that
occur. This was helpful for confirming that the IP address was set properly
since the call-id contains the IP address. It also will be helpful for
troubleshooting purposes when following a call in the debug logs.
(closes issue ASTERISK-20805)
Reported by: Bryan Hunt
Tested by: Bryan Hunt, Michael L. Young
Patches:
asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2255/
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r378585 | kmoore | 2013-01-04 16:19:16 -0600 (Fri, 04 Jan 2013) | 13 lines
Fix pjproject compilation in certain circumstances
On a fresh checkout of Asterisk 11, running make before ./configure
could cause the pjproject subdirectory to get in an odd state that
would prevent compilation. This patch by Tilghman prevents that from
occurring.
(closes issue ASTERISK-20681)
Patch-by: Tilghman Lesher
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r378557 | file | 2013-01-04 15:18:07 -0600 (Fri, 04 Jan 2013) | 11 lines
Don't pass STUN packets through the SRTP unprotect function.
(closes issue AST-1036)
Reported by: jbigelow
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r378543 | lathama | 2013-01-04 10:44:33 -0600 (Fri, 04 Jan 2013) | 6 lines
Doxygen Cleanups
Baseline clean up of formating to make room for extended documentation
(issue ASTERISK-20259)
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r378516 | elguero | 2013-01-03 16:14:20 -0600 (Thu, 03 Jan 2013) | 25 lines
Fix Queue Log Reporting Every Call COMPLETECALLER With "h" Extension Present
When the "h" extension is present within the context of the queue, all calls
are being reported COMPLETECALLER even when the agent is hanging up the call.
This patch checks to see if the agent hung-up or not instead of only relying on
checking if the queue (caller) channel hung-up or not. It would appear that
having the h extension in the mix, the pbx goes to the h extension,
"hanging-up" the queue channel and triggering the reporting of COMPLETECALLER.
(closes issue ASTERISK-20743)
Reported by: call
Tested by: call, Michael L. Young
Patches:
asterisk-20743-q-cmplt-caller.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2256/
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r378488 | rmudgett | 2013-01-03 13:42:54 -0600 (Thu, 03 Jan 2013) | 15 lines
chan_agent: Fix wrapup time wait response.
* Made agent_cont_sleep() and agent_ack_sleep() stop waiting if the wrapup
time expires. agent_cont_sleep() had tried but returned the wrong value
to stop waiting.
* Made agent_ack_sleep() take a struct agent_pvt pointer instead of a void
pointer for better type safety.
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r378458 | rmudgett | 2013-01-03 12:47:29 -0600 (Thu, 03 Jan 2013) | 18 lines
chan_agent: Misc code cleanup.
* Fix off-nominal path resource cleanup in agent_request().
* Create agent_pvt_destroy() to eliminate inlined versions in many places.
* Pull invariant code out of loop in add_agent().
* Remove redundant module user references in login_exec().
* Remove unused struct agent_pvt logincallerid[] member.
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r378460 | kmoore | 2013-01-03 12:51:43 -0600 (Thu, 03 Jan 2013) | 13 lines
Add missing test event
This test event was missing from channel.c causing the dial_LS_options
test to fail intermittently because of a race condition where most code
paths emitted the test event but this one did not. The dial_LS_options
test should stop bouncing now.
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r378429 | rmudgett | 2013-01-03 11:48:14 -0600 (Thu, 03 Jan 2013) | 10 lines
chan_agent: Fix agent_indicate() locking.
Avoid deadlock potential with local channels and simplify the locking.
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r378410 | mjordan | 2013-01-03 09:37:31 -0600 (Thu, 03 Jan 2013) | 13 lines
Prevent crashes in res_xmpp when receiving large messages
Similar to r378287, res_xmpp was marshaling data read from an external source
onto the stack. For a sufficiently large message, this could cause a stack
overflow. This patch modifies res_xmpp in a similar fashion to res_jabber by
removing the stack allocation, as it was unnecessary.
(issue ASTERISK-20658)
Reported by: wdoekes
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r378412 | file | 2013-01-03 09:40:21 -0600 (Thu, 03 Jan 2013) | 11 lines
Prevent exhaustion of system resources through exploitation of event cache
This patch changes res_xmpp to no longer cache events under certain circumstances.
(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
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r378414 | tilghman | 2013-01-03 10:04:11 -0600 (Thu, 03 Jan 2013) | 11 lines
Add aliases to the Directory.
This is an interesting feature that allows additional strings to be used to
search the Directory, primarily intended to be used with nicknames, but could
be used with affiliations and the like. Because the name field is used in
more than one place (such as email notifications), it is important that these
additional strings not be placed in the name field, but be specified
separately.
Review: https://reviewboard.asterisk.org/r/2244/
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r378374 | rmudgett | 2013-01-02 15:23:16 -0600 (Wed, 02 Jan 2013) | 33 lines
Fix AMI redirect action with two channels failing to redirect both channels.
The AMI redirect action can fail to redirect two channels that are bridged
together. There is a race between the AMI thread redirecting the two
channels and the bridge thread noticing that a channel is hungup from the
redirects.
* Made the bridge wait for both channels to be redirected before exiting.
* Made the AMI redirect check that all required headers are present before
proceeding with the redirection.
* Made the AMI redirect require that any supplied ExtraChannel exist
before proceeding. Previously the code fell back to a single channel
redirect operation.
(closes issue ASTERISK-18975)
Reported by: Ben Klang
(closes issue ASTERISK-19948)
Reported by: Brent Dalgleish
Patches:
jira_asterisk_19948_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Thomas Sevestre, Deepak Lohani, Kayode
Review: https://reviewboard.asterisk.org/r/2243/
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r378377 | mjordan | 2013-01-02 16:10:32 -0600 (Wed, 02 Jan 2013) | 24 lines
Prevent crashes from occurring when reading from data sources with large values
When reading configuration data from an Asterisk .conf file or when pulling
data from an Asterisk RealTime backend, Asterisk was copying the data on the
stack for manipulation. Unfortunately, it is possible to read configuration
data or realtime data from some data source that provides a large blob of
characters. This could potentially cause a crash via a stack overflow.
This patch prevents large sets of data from being read from an ARA backend or
from an Asterisk conf file.
(issue ASTERISK-20658)
Reported by: wdoekes
Tested by: wdoekes, mmichelson
patches:
* issueA20658_dont_process_overlong_config_lines.patch uploaded by wdoekes (license 5674)
* issueA20658_func_realtime_limit.patch uploaded by wdoekes (license 5674)
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r378384 | mjordan | 2013-01-02 16:19:32 -0600 (Wed, 02 Jan 2013) | 11 lines
Clean up app_mysql's application entry points to properly parse arguments
When parsing arguments, application entry points should not attempt to
directly modify the parameters to the function. This patch properly duplicates
the passed in parameters before attempting to parse them.
(issue ASTERISK-20658)
Reported by: wdoekes
patches:
issueA20658_sanitize_app_mysql.patch uploaded by wdoekes (license 5674)
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r378322 | mjordan | 2013-01-02 12:11:59 -0600 (Wed, 02 Jan 2013) | 33 lines
Prevent exhaustion of system resources through exploitation of event cache
Asterisk maintains an internal cache for devices in the event subsystem. The
device state cache holds the state of each device known to Asterisk, such that
consumers of device state information can query for the last known state for
a particular device, even if it is not part of an active call. The concept of
a device in Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls are allowed
in Asterisk. A device was automatically created and stored in the cache for
each anonymous call that occurred; this was possible in the SIP and IAX2
channel drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
are never removed from the system, allowing anonymous calls to potentially
exhaust a system's resources.
This patch changes the event cache subsystem and device state management to
no longer cache devices that are not associated with a physical entity.
(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
patches:
event-cachability-3.diff uploaded by jcolp (license 5000)
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r378288 | mjordan | 2013-01-02 09:39:42 -0600 (Wed, 02 Jan 2013) | 36 lines
Resolve crashes due to large stack allocations when using TCP
Asterisk had several places where messages received over various network
transports may be copied in a single stack allocation. In the case of TCP,
since multiple packets in a stream may be concatenated together, this can
lead to large allocations that overflow the stack.
This patch modifies those portions of Asterisk using TCP to either
favor heap allocations or use an upper bound to ensure that the stack will not
overflow:
* For SIP, the allocation now has an upper limit
* For HTTP, the allocation is now a heap allocation instead of a stack
allocation
* For XMPP (in res_jabber), the allocation has been eliminated since it was
unnecesary.
Note that the HTTP portion of this issue was independently found by Brandon
Edwards of Exodus Intelligence.
(issue ASTERISK-20658)
Reported by: wdoekes, Brandon Edwards
Tested by: mmichelson, wdoekes
patches:
ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049)
issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674)
issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674)
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r378259 | lathama | 2013-01-01 13:02:52 -0600 (Tue, 01 Jan 2013) | 5 lines
Add UUID packages now required to configure
In ASTERISK-20726 UUID was added to Asterisk. This commit is to add the dependancies to the install script
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r378248 | seanbright | 2013-01-01 11:03:59 -0600 (Tue, 01 Jan 2013) | 2 lines
Bail out early when building an ast_trans_pvt and the translator doesn't supply a 'newpvt'
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r378249 | seanbright | 2013-01-01 11:10:42 -0600 (Tue, 01 Jan 2013) | 2 lines
Revert 378248. I changed the logic of this function unitentionally, pointed out by file.
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r378220 | kmoore | 2012-12-31 08:46:06 -0600 (Mon, 31 Dec 2012) | 18 lines
Ensure chan_sip rejects encrypted streams without crypto info
This ensures that Asterisk rejects encrypted media streams (RTP/SAVP
audio and video) that are missing cryptographic keys and ensures that
the incoming SDP is consistent with RFC4568 as far as having a crypto
attribute present for any SAVP streams.
Review: https://reviewboard.asterisk.org/r/2204/
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r378166 | rmudgett | 2012-12-20 15:51:03 -0600 (Thu, 20 Dec 2012) | 8 lines
Give the causes[] a struct name.
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r378122 | kmoore | 2012-12-18 11:48:36 -0600 (Tue, 18 Dec 2012) | 17 lines
Add test events for time limit-related hangups
This patch adds hangup-related test events in order to support testing
of time-limited bridges. This aids in testing the S() and L() bridge
options.
(issue SWP-4713)
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r378091 | rmudgett | 2012-12-17 17:02:54 -0600 (Mon, 17 Dec 2012) | 22 lines
Make chan_local module references tied to local_pvt lifetime.
The chan_local module references were manually tied to the existence of
the ;1 and ;2 channel links.
* Made chan_local module references tied to the existence of the local_pvt
structure as well as automatically take care of the module references.
* Tweaked the wording of the local_fixup() failure warning message to make
sense.
Review: https://reviewboard.asterisk.org/r/2181/
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r378095 | rmudgett | 2012-12-17 17:10:42 -0600 (Mon, 17 Dec 2012) | 11 lines
Fix potential double free when unloading a module.
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r378081 | rmudgett | 2012-12-17 15:22:21 -0600 (Mon, 17 Dec 2012) | 7 lines
chan_local: Parse dial string consistently.
* Fix local_alloc() unexpected limitation of exten and context length from
a combined length of 80 characters to a normal 80 characters each.
* Made local_alloc() and local_devicestate() parse the same way.
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r378072 | rmudgett | 2012-12-17 14:34:25 -0600 (Mon, 17 Dec 2012) | 9 lines
chan_local: Misc lock and ref tweaks.
* awesome_locking() does not need to thrash the pvt lock as much.
* local_setoption() does not need to check for NULL pvt on cleanup since
it will never be NULL.
* Made ref the pvt before locking for consistency.
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r378074 | qwell | 2012-12-17 14:59:51 -0600 (Mon, 17 Dec 2012) | 10 lines
Make libasteriskssl.so symlink use a relative path.
This was causing issues when using DESTDIR, since the path to which the link
pointed is not likely to exist (and not useful to exist) on the target system.
(issue ASTNOW-284)
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r378063 | jrose | 2012-12-14 16:34:18 -0600 (Fri, 14 Dec 2012) | 8 lines
Features: BRIDGE_FEATURES variable automixmonitor support and use proper party
BRIDGE_FEATURES did not previously support the automixmonitor feature. Now it
does. In addition, the BRIDGE_FEATURES variable would not apply features to
the proper party based on whether the feature option letter was in caps or
in lowercase (both ways would apply it to the caller). Now uppercase applies
to the caller while lowercase applies to the callee (like with the dial option)
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r378064 | rmudgett | 2012-12-14 16:45:03 -0600 (Fri, 14 Dec 2012) | 4 lines
chan_agent: Remove some duplicated code.
No need to check for an agent twice. Santa does that.
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r378039 | rmudgett | 2012-12-14 15:35:44 -0600 (Fri, 14 Dec 2012) | 26 lines
app_queue: Revert bad ringinuse=no patch.
With the option ringinuse=no set, the patch committed for ASTERISK-16115
causes non-SIP queue members to never be called because the device state
is checked after a channel is created to determine if the member is busy.
These queue members always get the "Member %s is busy, cannot dial"
message.
Most channel drivers other than chan_sip use the default device state
handling. The default device-state state is considered in use or unknown
if the channel exists or not respectively.
(closes issue ASTERISK-20801)
Reported by: rmudgett
Patches:
jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621) patch uploaded by rmudgett
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r378029 | rmudgett | 2012-12-14 14:22:36 -0600 (Fri, 14 Dec 2012) | 1 line
app_queue: Make update_status() not return anything.
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r378011 | wedhorn | 2012-12-13 19:55:43 -0600 (Thu, 13 Dec 2012) | 15 lines
Fix skinny to recognise vmexten in general section of conf
Fixup the vmexten so if globally set in general section will be honored by
chan_skinny. Also get rid of the 'global_' part of variable name to match
regexten.
(closes issue AST-20790)
Reported by: snuffy
Tested by: snuffy, myself
Patches:
skinny-vm.diff uploaded by snuffy (license 5024)
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r378006 | wedhorn | 2012-12-13 19:02:15 -0600 (Thu, 13 Dec 2012) | 8 lines
Add g722 codec support to skinny
(closes issue AST-20788)
Reported by: snuffy
Tested by: snuffy, myself
Patches:
skinny-g722.diff uploaded by snuffy (license 5024)
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r378000 | seanbright | 2012-12-13 15:20:32 -0600 (Thu, 13 Dec 2012) | 8 lines
Make generate_exchange_uuid() always return the passed ast_str pointer.
I changed this code earlier to return NULL if it wasn't able to generate a UUID,
whereas the earlier code would always return the ast_str that was passed in.
Switch back to returning the ast_str, only set it to the empty string instead if
UUID generation fails. We still do a validity check later which will catch this
and blow up if necessary.
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r378001 | wedhorn | 2012-12-13 15:25:31 -0600 (Thu, 13 Dec 2012) | 9 lines
Minor fixes for chan_skinny
Whitespace, change SUBSTATE_ONHOOK to correct SKINNY_ONHOOK and
correct len of 2 strcmp in skinny_setdebug(). (see opticron's review
on https://reviewboard.asterisk.org/r/2240/)
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r378002 | rmudgett | 2012-12-13 15:28:15 -0600 (Thu, 13 Dec 2012) | 35 lines
confbridge: Fix MOH on simultaneous user entry to a new conference.
When two users entered a new conference simultaneously, one of the callers
hears MOH. This happened if two unmarked users entered simultaneously and
also if a waitmarked and a marked user entered simultaneously.
* Created a confbridge internal MOH API to eliminate the inlined MOH
handling code. Note that the conference mixing bridge needs to be locked
when actually starting/stopping MOH because there is a small window
between the conference join unsuspend MOH and actually joining the mixing
bridge.
* Created the concept of suspended MOH so it can be interrupted while
conference join announcements to the user and DTMF features can operate.
* Suspend any MOH until the user is about to actually join the mixing
bridge of the conference. This way any pre-join file playback does not
need to worry about MOH.
* Made post-join actions only play deferred entry announcement files.
Changing the user/conference state during that time is not protected or
controlled by the state machine.
(closes issue ASTERISK-20606)
Reported by: Eugenia Belova
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2232/
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r377994 | dlee | 2012-12-13 15:15:44 -0600 (Thu, 13 Dec 2012) | 1 line
Fixed svn merge property breakage from r377986
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r377986 | wedhorn | 2012-12-13 12:28:41 -0600 (Thu, 13 Dec 2012) | 14 lines
Fix skinny debug tab completion
Review the syntax of the 'skinny debug' command to show more than
just 'show' for options to 'skinny debug' command.
(closes issue ASTERISK-20789)
Reported by: snuffy
Tested by: snuffy, myself
Patches:
skinny-debug.diff uploaded by snuffy (license 5024)
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r377981 | dlee | 2012-12-13 10:43:40 -0600 (Thu, 13 Dec 2012) | 1 line
Bail configure if it can't find libuuid.
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r377966 | kmoore | 2012-12-13 08:28:57 -0600 (Thu, 13 Dec 2012) | 23 lines
Ensure Min-SE is included in outbound INVITEs
Asterisk now includes Min-SE in outbound INVITEs when the value is not
90 (the default) and session timers are not disabled. This has the
effect of Asterisk following RFC4028 more closely with regard to 422
responses and preventing situations in which Asterisk would be forced
to temporarily accept a call to tear it down based on a Session-Expires
below the locally configured Min-SE.
(issue SWP-5051)
Review: https://reviewboard.asterisk.org/r/2222/
Reported-by: Kinsey Moore
Patch-by: Kinsey Moore
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r377925 | newtonr | 2012-12-12 16:43:40 -0600 (Wed, 12 Dec 2012) | 18 lines
Incremented EXTRA_SOUNDS_VERSION in sounds/Makefile to 1.4.12 for new Extra Sounds releases
See CHANGES-* files in English extra 1.4.12 tarballs for new sound prompts added.
(closes ASTERISK-20328)
Reported by: Matt Jordan
(closes AST-755)
Reported by: John Bigelow
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r377915 | elguero | 2012-12-11 22:43:18 -0600 (Tue, 11 Dec 2012) | 17 lines
Convert Dynamic Features Buffer To Use ast_str
Currently, the buffer for the dynamic features list is set to a fixed size of
128. If the list is bigger than that, it results in the dynamic feature(s) not
being recognized.
This patch changes the buffer from a fixed size to a dynamic one.
(closes issue ASTERISK-20680)
Reported by: Clod Patry
Tested by: Michael L. Young
Patches:
asterisk-20680-dynamic-features-v2.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2221/
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r377911 | mmichelson | 2012-12-11 18:02:31 -0600 (Tue, 11 Dec 2012) | 22 lines
Fix a potential deadlock in chan_sip during transfers.
The issue comes from the fact that transfers may perform
a redirecting update on a channel. The issue is that lock
inversion between the channel and its tech_pvt occurs since
the channel lock is released during the transfer process.
The fix is to move when the redirecting update occurs to a
place where neither the tech_pvt or the channel is locked so
that the two can be locked in the proper order.
(closes issue ASTERISK-20708)
reported by Mark Michelson
patches:
ASTERISK-20708-3.patch uploaded by Mark Michelson (License #5049)
Tested by:
Tim Ringenbach at Asteria Solutions Group
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r377906 | mmichelson | 2012-12-11 16:42:11 -0600 (Tue, 11 Dec 2012) | 3 lines
Add test events necessary for bridging tests to be able to properly run.
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r377809 | rmudgett | 2012-12-11 14:05:32 -0600 (Tue, 11 Dec 2012) | 23 lines
Cleanup pbx on exit.
* Cleanup CLI commands on exit.
* Unreference hints and statecbs containers on exit.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
pbx-cleanup-1_8.patch (license #5909) patch uploaded by Corey Farrell
pbx-cleanup-10.patch (license #5909) patch uploaded by Corey Farrell
pbx-cleanup-11-trunk.patch (license #5909) patch uploaded by Corey Farrell
Modified
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Some general cleanup, plus we now send state changes when threads activate.
This is now ready for review board, imo!
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Mark Michelson [Tue, 11 Dec 2012 16:34:00 +0000 (16:34 +0000)]
Add auto-increment option and accompanying test.
This allows for the threadpool to automatically grow if tasks
are pushed to it and no idle threads are currently available.
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Mark Michelson [Tue, 11 Dec 2012 04:23:31 +0000 (04:23 +0000)]
Some documentation fixes and function call name fixes.
The documentation for taskprocessors was incorrect with
regards to when a listener's alloc callback was called.
I also made the names of queued function calls in the
threadpool more uniform.
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r377774 | rmudgett | 2012-12-10 20:44:53 -0600 (Mon, 10 Dec 2012) | 19 lines
Cleanup logger on exit.
* Cleanup CLI commands, destroy verbosers and logchannels lists on exit.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
logger-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell
Modified
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r377743 | rmudgett | 2012-12-10 20:13:37 -0600 (Mon, 10 Dec 2012) | 25 lines
Cleanup indications on exit.
* Made ast_unregister_indication_country() unlink the found tone zone
before selecting a new default_tone_zone to make it impossible to select
the tone zone being unregistered again.
* Ringcadence is no longer parsed twice in store_config_tone_zone().
* Cleanup CLI commands and destroy default_tone_zone on exit.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
indications-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell
Modified
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r377707 | rmudgett | 2012-12-10 18:37:01 -0600 (Mon, 10 Dec 2012) | 20 lines
Cleanup dnsmgr on exit.
* Cleanup dnsmgr thread and CLI commands on exit.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
dnsmgr-cleanup-1_8.patch (license #5909) patch uploaded by Corey Farrell
dnsmgr-cleanup-10-11-trunk.patch (license #5909) patch uploaded by Corey Farrell
Modified
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r377711 | rmudgett | 2012-12-10 19:04:50 -0600 (Mon, 10 Dec 2012) | 19 lines
Cleanup event on exit.
* Cleanup CLI commands on exit.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
event_shutdown-10-only.patch (license #5909) patch uploaded by Corey Farrell
event_shutdown-1_8-11-trunk.patch (license #5909) patch uploaded by Corey Farrell
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Mark Michelson [Mon, 10 Dec 2012 20:14:23 +0000 (20:14 +0000)]
Solve the issue about the "CHANGE THIS" and "XXX CHANGE THIS XXX" taskprocessor names.
Unfortunately, this required a taskprocessor listener change that makes listener allocation
utterly silly. I'm going to change the scheme so that allocation of taskprocessor listeners
is done internally within taskprocessor code. This will make it parallel with threadpool
code, which is a good thing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377687
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Mark Michelson [Mon, 10 Dec 2012 19:32:50 +0000 (19:32 +0000)]
Make thread state waiting a little less ugly and not possible to lock up.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377686
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r377658 | kmoore | 2012-12-10 10:56:37 -0600 (Mon, 10 Dec 2012) | 20 lines
Ensure ReceiveFax provides a CED tone via T.38
When using res_fax_digium, the T.38 CED tone was not being provided
properly which would cause some incoming faxes to fail. This was not an
issue with res_fax_spandsp since it does not strictly honor the
send_ced flag and sends the CED tone whenever receiving a T.38 fax.
(closes issue FAX-343)
Reported-by: Benjamin Tietz
Patch-by: Kinsey Moore
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Mark Michelson [Mon, 10 Dec 2012 15:55:56 +0000 (15:55 +0000)]
Improve timedwaits in tests to actually behave like they should.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377654
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r377626 | kmoore | 2012-12-10 08:45:52 -0600 (Mon, 10 Dec 2012) | 20 lines
Handle Session-Expires less than local Min-SE in 200 OK
Ensure that a call is immediately torn down if a Session-Expires value
received in a 200 OK is less than the local Min-SE. This also prevents
Asterisk from allowing calls with Session-Expires below the
RFC4028-mandated minimum (90s).
(closes issue ASTERISK-20653)
Review: https://reviewboard.asterisk.org/r/2237/
Patch-by: Kinsey Moore
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r377594 | igorg | 2012-12-10 00:56:04 -0600 (Mon, 10 Dec 2012) | 15 lines
Fix codec mismatch
Fix code to send in both rx and tx open stream messages correct codecs. Found that on phase 0/1 phones wrong codecs cause to no audio in some situations.
(issue ASTERISK-20183)
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r377595 | igorg | 2012-12-10 01:03:48 -0600 (Mon, 10 Dec 2012) | 3 lines
Add firmware information to CLI devices listing
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r377579 | igorg | 2012-12-09 23:29:04 -0600 (Sun, 09 Dec 2012) | 7 lines
Remove trailing whitespaces in number from incoming redial list.
Reported by: Igor Olhovskiy
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Mark Michelson [Mon, 10 Dec 2012 06:13:09 +0000 (06:13 +0000)]
Add threadpool options and accompanying test.
The only test added so far is an idle thread timeout
option. This will greatly aid threadpool users who wish
to maintain a threadpool by allowing for idle threads to
die out as necessary.
Test passes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377580
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Mark Michelson [Mon, 10 Dec 2012 05:25:38 +0000 (05:25 +0000)]
Improve shutdown procedure.
This helps tests to pass more often than before.
They are far less likely to queue extra processes
into the control taskprocessor since they are prevented
once the threadpool begins to shut down.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377578
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Mark Michelson [Mon, 10 Dec 2012 04:08:29 +0000 (04:08 +0000)]
Add safeguards to ensure we don't improperly access a destroyed taskprocessor.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377556
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r377506 | tilghman | 2012-12-09 19:29:32 -0600 (Sun, 09 Dec 2012) | 11 lines
Remove some dead code and additionally handle a case that wasn't handled.
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r377512 | tilghman | 2012-12-09 19:41:50 -0600 (Sun, 09 Dec 2012) | 22 lines
Improve documentation by making all of the colors used readable,
no matter what the background color is.
Dark blue on a black background is unreadable, as is yellow on a
light background. This patch turns on the bright attribute for
colors when on a dark background and turns *off* the bright
attribute when the -W command line option is used (indicating a
_light_ background). This ensures that text is readable in both
cases.
Patch by: tilghman
Review: https://reviewboard.asterisk.org/r/2224
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Mark Michelson [Sun, 9 Dec 2012 19:10:14 +0000 (19:10 +0000)]
Remove unnecessary debugging and add some useful debugging.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377476
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Mark Michelson [Sun, 9 Dec 2012 19:06:35 +0000 (19:06 +0000)]
Add some useful debugging in threadpool test failure conditions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377475
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Mark Michelson [Sun, 9 Dec 2012 18:56:25 +0000 (18:56 +0000)]
Create longer thread destruction test.
This one involves shrinking the threadpool in such
a way that both idle and active threads are affected.
This test made me re-realize why the zombie state exists,
so I re-added it. We don't want to clog up the control
taskprocessor by waiting on active threads to complete
what they are doing. Instead, we mark them as zombies so
that when they are done, they can clean themselves up
properly.
Without the zombie state available, the new test actually
will deadlock.
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r377463 | file | 2012-12-08 19:23:44 -0600 (Sat, 08 Dec 2012) | 9 lines
Add missing support for "who hung up" to chan_motif.
(closes issue ASTERISK-20671)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2208/
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r377434 | rmudgett | 2012-12-07 18:30:40 -0600 (Fri, 07 Dec 2012) | 20 lines
Fix order of SIP allow/disallow in MySQL contrib script.
Using the contrib sippeers.sql script to create the sippeers MySQL table
would result in being unable to place calls if you set the disallow value
to all.
(closes issue ASTERISK-20756)
Reported by: Andre Luis
Patches:
sippeers.patch patch uploaded by Andre Luis
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Mark Michelson [Sat, 8 Dec 2012 00:22:33 +0000 (00:22 +0000)]
Add a task distribution test.
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r377402 | rmudgett | 2012-12-07 17:45:15 -0600 (Fri, 07 Dec 2012) | 11 lines
MALLOC_DEBUG: Only wait if we want atexit allocation dumps.
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Mark Michelson [Fri, 7 Dec 2012 23:42:42 +0000 (23:42 +0000)]
Add threadpool reactivation test.
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Mark Michelson [Fri, 7 Dec 2012 22:35:11 +0000 (22:35 +0000)]
Add a single-thread multi-task test.
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