mirror of https://github.com/asterisk/asterisk
master
22
23
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${ noResults }
23272 Commits (bdd8da406bdce8dcb92104f8bcbbb8c42f57646d)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
bdd8da406b |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378652 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
|
ad73fe2c9f |
Merged revisions 378634 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ........ 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/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378639 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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file:///srv/subversion/repos/asterisk/trunk ................ 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/ ........ Merged revisions 378622 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ 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/ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378629 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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file:///srv/subversion/repos/asterisk/trunk ................ 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 ........ Merged revisions 378591 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378592 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378600 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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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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file:///srv/subversion/repos/asterisk/trunk ................ 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 ........ Merged revisions 378553 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378555 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378560 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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file:///srv/subversion/repos/asterisk/trunk ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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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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file:///srv/subversion/repos/asterisk/trunk ................ 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. ........ Merged revisions 378486 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378487 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378495 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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file:///srv/subversion/repos/asterisk/trunk ................ 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. ........ Merged revisions 378456 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378457 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ 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. ........ Merged revisions 378455 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378459 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378467 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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file:///srv/subversion/repos/asterisk/trunk ................ 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. ........ Merged revisions 378427 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378428 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378436 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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file:///srv/subversion/repos/asterisk/trunk ................ 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 ........ Merged revisions 378409 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ 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 ........ Merged revisions 378411 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ 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/ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378422 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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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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file:///srv/subversion/repos/asterisk/trunk ........ 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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378264 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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file:///srv/subversion/repos/asterisk/trunk ........ 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' ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378254 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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file:///srv/subversion/repos/asterisk/trunk ................ 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/ ........ Merged revisions 378217 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378218 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378219 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378228 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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file:///srv/subversion/repos/asterisk/trunk ................ r378166 | rmudgett | 2012-12-20 15:51:03 -0600 (Thu, 20 Dec 2012) | 8 lines Give the causes[] a struct name. ........ Merged revisions 378164 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378165 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378169 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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file:///srv/subversion/repos/asterisk/trunk ................ 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) ........ Merged revisions 378119 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378120 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378121 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378126 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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file:///srv/subversion/repos/asterisk/trunk ................ 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/ ........ Merged revisions 378088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378089 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378090 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ r378095 | rmudgett | 2012-12-17 17:10:42 -0600 (Mon, 17 Dec 2012) | 11 lines Fix potential double free when unloading a module. ........ Merged revisions 378092 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378093 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378094 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378099 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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file:///srv/subversion/repos/asterisk/trunk ................ 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. ................ 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) ........ Merged revisions 378073 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378076 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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file:///srv/subversion/repos/asterisk/trunk ........ 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) ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378066 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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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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file:///srv/subversion/repos/asterisk/trunk ........ r378029 | rmudgett | 2012-12-14 14:22:36 -0600 (Fri, 14 Dec 2012) | 1 line app_queue: Make update_status() not return anything. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378031 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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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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Merged revisions 378000-378002 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ................ 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. ................ 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/) ........ Merged revisions 377991 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ 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/ ........ Merged revisions 377992 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 377993 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378003 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 377994 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ........ r377994 | dlee | 2012-12-13 15:15:44 -0600 (Thu, 13 Dec 2012) | 1 line Fixed svn merge property breakage from r377986 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377996 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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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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file:///srv/subversion/repos/asterisk/trunk ........ r377981 | dlee | 2012-12-13 10:43:40 -0600 (Thu, 13 Dec 2012) | 1 line Bail configure if it can't find libuuid. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377982 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Resolve conflict and reset automerge.
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automerge cancel
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file:///srv/subversion/repos/asterisk/trunk ................ 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 ........ Merged revisions 377946 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377947 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 377948 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377968 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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file:///srv/subversion/repos/asterisk/trunk ................ 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 ........ Merged revisions 377922 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377923 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 377924 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377928 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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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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file:///srv/subversion/repos/asterisk/trunk ................ 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 ........ Merged revisions 377910 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377912 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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file:///srv/subversion/repos/asterisk/trunk ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377907 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Reset automerge property.
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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! git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377805 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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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. git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377803 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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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. git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377802 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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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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