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Author	SHA1	Message	Date
Automerge script	1db640cdbc	Merged revisions 372090 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r372090 | mmichelson | 2012-08-30 15:53:09 -0500 (Thu, 30 Aug 2012) | 17 lines Prevent crash on shutdown due to refcount error on queues container. When app_queue is unloaded, the queues container has its refcount decremented, potentially to 0. Then the taskprocessor responsible for handling device state changes is unreferenced. If the taskprocessor happens to be just about to run its task, then it will create and destroy an iterator on the queues container. This can cause the refcount on the queues container to increase to 1 and then back to 0. Going back to 0 a second time results in double frees. This failure was seen periodically in the testsuite when Asterisk would shut down. ........ Merged revisions 372089 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@372115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	415082bc75	Merged revisions 372049 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r372049 | mmichelson | 2012-08-30 13:33:37 -0500 (Thu, 30 Aug 2012) | 16 lines Help prevent ringing queue members from being rung when ringinuse set to no. Queue member status would not always get updated properly when the member was called, thus resulting in the member getting multiple calls. With this change, we update the member's status at the time of calling, and we also check to make sure the member is still available to take the call before placing an outbound call. (closes issue ASTERISK-16115) reported by nik600 Patches: app_queue.c-svn-r370418.patch uploaded by Italo Rossi (license #6409) ........ Merged revisions 372048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@372088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	4e0efef6eb	Merged revisions 371999,372020 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371999 | mjordan | 2012-08-30 11:06:47 -0500 (Thu, 30 Aug 2012) | 36 lines AST-2012-012: Resolve AMI User Unauthorized Shell Access through ExternalIVR The AMI Originate action can allow a remote user to specify information that can be used to execute shell commands on the system hosting Asterisk. This can result in an unwanted escalation of permissions, as the Originate action, which requires the "originate" class authorization, can be used to perform actions that would typically require the "system" class authorization. Previous attempts to prevent this permission escalation (AST-2011-006, AST-2012-004) have sought to do so by inspecting the names of applications and functions passed in with the Originate action and, if those applications/functions matched a predefined set of values, rejecting the command if the user lacked the "system" class authorization. As reported by IBM X-Force Research, the "ExternalIVR" application is not listed in the predefined set of values. The solution for this particular vulnerability is to include the "ExternalIVR" application in the set of defined applications/functions that require "system" class authorization. Unfortunately, the approach of inspecting fields in the Originate action against known applications/functions has a significant flaw. The predefined set of values can be bypassed by creative use of the Originate action or by certain dialplan configurations, which is beyond the ability of Asterisk to analyze at run-time. Attempting to work around these scenarios would result in severely restricting the applications or functions and prevent their usage for legitimate means. As such, any additional security vulnerabilities, where an application/function that would normally require the "system" class authorization can be executed by users with the "originate" class authorization, will not be addressed. Instead, the README-SERIOUSLY.bestpractices.txt file has been updated to reflect that the AMI Originate action can result in commands requiring the "system" class authorization to be executed. Proper system configuration can limit the impact of such scenarios. (closes issue ASTERISK-20132) Reported by: Zubair Ashraf of IBM X-Force Research ........ Merged revisions 371998 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r372020 | mjordan | 2012-08-30 11:22:54 -0500 (Thu, 30 Aug 2012) | 17 lines AST-2012-013: Resolve ACL rules being ignored during calls by some IAX2 peers When an IAX2 call is made using the credentials of a peer defined in a dynamic Asterisk Realtime Architecture (ARA) backend, the ACL rules for that peer are not applied to the call attempt. This allows for a remote attacker who is aware of a peer's credentials to bypass the ACL rules set for that peer. This patch ensures that the ACLs are applied for all peers, regardless of their storage mechanism. (closes issue ASTERISK-20186) Reported by: Alan Frisch Tested by: mjordan, Alan Frisch ........ Merged revisions 372015 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@372027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	c94343a893	Merged revisions 371962 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371962 | mjordan | 2012-08-30 07:48:07 -0500 (Thu, 30 Aug 2012) | 17 lines Restore CODING-GUIDELINES to doc folder In r294740, the CODING-GUIDELINES was removed from the doc folder in favor of the content on the Asterisk wiki. Some folks still look in the doc folder initially for coding guideline suggestions; as such, this patch adds a CODING-GUIDELINES file back into the doc folder. The content of the file merely points to the correct page on the Asterisk wiki where the coding guidelines currently live. (closes issue ASTERISK-20279) Reported by: Andrew Latham Patches: CODING-GUIDELINES.diff uploaded by Andrew Latham (license 5985) ........ Merged revisions 371961 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371988 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	33c5db3259	Merged revisions 371920 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371920 | jrose | 2012-08-29 15:58:21 -0500 (Wed, 29 Aug 2012) | 5 lines app_meetme: Adding test events for following activity in MeetMe. ........ Merged revisions 371919 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	12ef471889	Merged revisions 371890 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371890 | rmudgett | 2012-08-29 14:40:20 -0500 (Wed, 29 Aug 2012) | 13 lines Initialize file descriptors for dummy channels to -1. Dummy channels usually aren't read from, but functions like SHELL and CURL use autoservice on the channel. (closes issue ASTERISK-20283) Reported by: Gareth Palmer Patches: svn-371580.patch (license #5169) patch uploaded by Gareth Palmer (modified) ........ Merged revisions 371888 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371918 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	57a546d181	Merged revisions 371861 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371861 | rmudgett | 2012-08-29 13:24:54 -0500 (Wed, 29 Aug 2012) | 15 lines Fix hangup cause passthrough regression. The v1.8 -r369258 change to fix the F and F(x) action logic introduced a regression in passing the hangup cause from the called channel to the caller channel. (closes issue ASTERISK-20287) Reported by: Konstantin Suvorov Patches: app_dial_hangupcause.patch (license #6421) patch uploaded by Konstantin Suvorov (modified) Tested by: rmudgett ........ Merged revisions 371860 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	67a1102114	Merged revisions 371825 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371825 | jrose | 2012-08-29 12:07:35 -0500 (Wed, 29 Aug 2012) | 8 lines chan_sip: Send 408 on retransmit timeout instead of 603 (closes issue ASTERISK-20124) Reported by: Walter Doekes ........ Merged revisions 371824 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	e10c8f2b5f	Merged revisions 371783,371789 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371783 | mmichelson | 2012-08-27 16:29:29 -0500 (Mon, 27 Aug 2012) | 9 lines Fix incorrect documentation of the MailboxStatus manager command. The "Waiting" field was misdocumented as reporting the number of messages waiting. In reality, it simply indicated the presence or absence of waiting messages. ........ Merged revisions 371782 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r371789 | mmichelson | 2012-08-27 16:49:51 -0500 (Mon, 27 Aug 2012) | 13 lines Fix misleading documentation in agents.conf.sample regarding ackcall usage. The documentation made it sound as if the DTMF acknowledgment was needed at the time the agent logs in, rather than when the agent is called. This is likely a relic from the days when there were multiple ways of logging in agents. (closes issue AST-962) reported by Steve Pitts ........ Merged revisions 371787 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	49057c85fc	Merged revisions 371748 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371748 | mmichelson | 2012-08-27 12:36:43 -0500 (Mon, 27 Aug 2012) | 10 lines Fix incorrectly documented option in queues.conf sharedlastcall defaults to "no" not "yes" (closes issue AST-979) reported by Steve Pitts ........ Merged revisions 371747 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	e948702e4d	Merged revisions 371719 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371719 | dlee | 2012-08-27 11:43:09 -0500 (Mon, 27 Aug 2012) | 15 lines Fixes ast_rwlock_timed[rd|wr]lock for BSD and variants. The original implementations simply wrap pthread functions, which take absolute time as an argument. The spinlock version for systems without those functions treated the argument as a delta. This patch fixes the spinlock version to be consistent with the pthread version. (closes issue ASTERISK-20240) Reported by: Egor Gorlin Patches: lock.c.patch uploaded by Egor Gorlin (license 6416) ........ Merged revisions 371718 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371746 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	b275e7d8b5	Merged revisions 371691 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371691 | kmoore | 2012-08-27 08:57:10 -0500 (Mon, 27 Aug 2012) | 14 lines Implement workaround for BETTER_BACKTRACES crash When compiling with BETTER_BACKTRACES enabled, Asterisk will sometimes crash when "core show locks" is run. This happens regularly in the testsuite since several tests run "core show locks" to help with debugging. This seems to be a fault with libraries on certain operating systems (notably CentOS 6.2/6.3) running on virtual machines and utilizing gcc 4.4.6. (closes issue ASTERISK-20090) ........ Merged revisions 371690 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	54927e840a	Merged revisions 371663 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371663 | alecdavis | 2012-08-26 18:06:14 -0500 (Sun, 26 Aug 2012) | 5 lines mf_detect: incorrectly used DTMF_GSIZE instead of MF_GSIZE ........ Merged revisions 371662 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	30708a6a2c	Merged revisions 371591 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371591 | mmichelson | 2012-08-21 15:40:18 -0500 (Tue, 21 Aug 2012) | 22 lines Fix misuses of asprintf throughout the code. This fixes three main issues * Change asprintf() uses to ast_asprintf() so that it pairs properly with ast_free() and no longer causes MALLOC_DEBUG to freak out. * When ast_asprintf() fails, set the pointer NULL if it will be referenced later. * Fix some memory leaks that were spotted while taking care of the first two points. (Closes issue ASTERISK-20135) reported by Richard Mudgett Review: https://reviewboard.asterisk.org/r/2071 ........ Merged revisions 371590 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371617 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	208299a943	Merged revisions 371545 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371545 | kmoore | 2012-08-20 10:27:15 -0500 (Mon, 20 Aug 2012) | 15 lines Ignore recovered zero-length secondary UDPTL packets In some cases, recovering lost packets using the secondary packet recovery mechanism with UDPTL/T.38 can result in the recovery of zero-length packets. These must be ignored or the frame generated from them can cause segfaults and allocation failures. (closes issue ASTERISK-19762) (closes issue ASTERISK-19373) Reported-by: Benjamin (bulkorok) Reported-by: Rob Gagnon (rgagnon) ........ Merged revisions 371544 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	581e938122	Merged revisions 371529 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r371529 | mjordan | 2012-08-17 21:34:10 -0500 (Fri, 17 Aug 2012) | 7 lines Remove old debug code from http configuration loading (closes issue ASTERISK-20254) Reported by: Andrew Latham Patches: http.diff uploaded by Andrew Latham (license #5985) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	29871c0945	Merged revisions 371491 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371491 | mjordan | 2012-08-17 15:21:30 -0500 (Fri, 17 Aug 2012) | 17 lines Fix memory leak in XML documentation When formatting documentation fields, the XML documentation parser calls xmldoc_get_formatted. This function allocates a string buffer at the beginning of its routine. Unfortunately, on certain code paths, it also calls xmldoc_string_cleanup, which assumes that it will create the string buffer. The previously allocated string buffer is then leaked by the xmldoc_string_cleanup routine. Now: we don't do that. (closes issue AST-932) Reported by: Alexander Homig ........ Merged revisions 371469 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	ae78f5d70b	Merged revisions 371437 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371437 | kmoore | 2012-08-17 10:51:06 -0500 (Fri, 17 Aug 2012) | 11 lines Add instrumentation to subsystem reloads When Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now generate TestEvent AMI events on subsystem reloads such as cdr, dnsmgr, extconfig, etc. (issue PQ-1126) ........ Merged revisions 371436 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	b2cd9a1b52	Merged revisions 371394,371398 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371394 | kmoore | 2012-08-16 17:42:53 -0500 (Thu, 16 Aug 2012) | 11 lines Add module reload instrumentation for TEST_FRAMEWORK This adds AMI events for module reloads when Asterisk is built with TEST_FRAMEWORK enabled and corrects generation of the module load AMI event. (issue PQ-1126) ........ Merged revisions 371393 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r371398 | twilson | 2012-08-16 17:50:12 -0500 (Thu, 16 Aug 2012) | 13 lines Handle integer over/under-flow in ast_parse_args The strtol family of functions will return *_MIN/*_MAX on overflow. To detect when an overflow has happened, errno must be set to 0 before calling the function, then checked afterward. (closes issue ASTERISK-20120) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2073/ ........ Merged revisions 371392 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	6660ea0ed1	Merged revisions 371358 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371358 | jrose | 2012-08-16 14:05:21 -0500 (Thu, 16 Aug 2012) | 11 lines chan_sip: Use pvt outgoing_call variable to set Remote-Party-ID Header Previously the pvt SIP_OUTGOING flag was used instead, which will frequently flip during reinvites. (closes issue AST-897) Reported by: Thomas Arimont ........ Merged revisions 371357 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	da652a5e68	Merged revisions 371338 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371338 | jrose | 2012-08-16 11:16:04 -0500 (Thu, 16 Aug 2012) | 14 lines chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK Under certain conditions, a SIP transaction involving directmedia wouldn't trigger a re-invite because the SDP answer was included in an ACK instead of in a message that we would have triggered the invite with. This patch just queues a source change control frame if the dialog is using directmedia when we find sdp for an ACK. (closes issue AST-913) Reported by: Thomas Arimont ........ Merged revisions 371337 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	0083779480	Merged revisions 371313 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371313 | mmichelson | 2012-08-15 18:19:09 -0500 (Wed, 15 Aug 2012) | 20 lines Fix bug where final queue member would not be removed from memory. If a static queue had realtime members, then there could be a potential for those realtime members not to be properly deleted from memory. If the queue's members were loaded from realtime and then all the members were deleted from the backend, then the queue would still think these members existed. The reason was that there was a short- circuit in code such that if there were no members found in the backend, then the queue would not be updated to reflect this. Note that this only affected static queues with realtime members. Realtime queues with realtime members were unaffected by this issue. (closes issue ASTERISK-19793) reported by Marcus Haas ........ Merged revisions 371306 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371323 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	b7631ef52b	Merged revisions 371271 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371271 | kmoore | 2012-08-15 15:15:08 -0500 (Wed, 15 Aug 2012) | 12 lines Avoid unconditional NULLing of mwipvt on relatedpeer on SIP dialog destruction The other instance of this bug was fixed by jcolp/file in r121496. If we are destroying a dialog only set the MWI dialog pointer on the related peer to NULL if it is the dialog currently being destroyed. (closes issue ASTERISK-20119) Patch-by: Misha Vodsedalek ........ Merged revisions 371270 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	c55f4e088d	Merged revisions 371250-371251 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r371250 | elguero | 2012-08-14 20:35:57 -0500 (Tue, 14 Aug 2012) | 17 lines Fix Segfault When Registering SIP Over WebSockets The helper function, get_address_family_filter, in chan_sip for dns resolution by address family was not recognizing the websockets transport and resulting in a null pointer being sent to functions in netsock2, in an attempt to determine if we are bound to ANY address ([::]) or not. This patch fixes this issue by handling the transport types SIP_TRANSPORT_WS and SIP_TRANSPORT_WSS which results in a sock address being set properly for use in determining the address family. (closes issue ASTERISK-20221) Reported by: Sven Beisiegel Tested by: Sven Beisiegel, James Mortensen Patches: asterisk-20221-ws-family-filter.diff uploaded by Michael L. Young (license 5026) ........ r371251 | elguero | 2012-08-14 20:43:23 -0500 (Tue, 14 Aug 2012) | 4 lines Reverting this change that was meant for branch 11. (issue ASTERISK-20221) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371254 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f7c4ca17f4	Merged revisions 371199,371203 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371199 | mmichelson | 2012-08-13 14:51:19 -0500 (Mon, 13 Aug 2012) | 5 lines Fix problem where incorrect pointer was checked for nullity. ........ Merged revisions 371198 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r371203 | kmoore | 2012-08-13 15:04:15 -0500 (Mon, 13 Aug 2012) | 13 lines Add test instrumentation This adds test instrumentation for loading and unloading of modules and for certain actions in MeetMe to be used in the testsuite or any other consumer of AMI events. These will only be generated when Asterisk is built with TEST_FRAMEWORK enabled. (issue PQ-1131) (issue PQ-1133) ........ Merged revisions 371201 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	36e8019901	Merged revisions 371142 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371142 | mmichelson | 2012-08-10 16:23:52 -0500 (Fri, 10 Aug 2012) | 18 lines Fix a couple of documentation problems in app_queue.c * The RemoveQueueMember app made mention of options that could be passed in, but no options are supported. I have removed the listing of options from the documentation. * The RQMSTATUS variable did not list "NOTDYNAMIC" as a possible value that could be set. (closes issue AST-949) reported by Steve Pitts (closes issue AST-954) reported by Steve Pitts ........ Merged revisions 371141 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	ba79302dad	Merged revisions 371090 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371090 | may | 2012-08-10 11:46:38 -0500 (Fri, 10 Aug 2012) | 12 lines remove ALREADYGONE flag on ooh323 call data by ooh323_indicate (CONGESTION/BUSY) due to call hasn't gone there really. This indication arrive from asterisk core not h.323 stack (closes issue ASTERISK-19308) Reported by: Dmitry Melekhov Patches: ASTERISK-19308.patch ........ Merged revisions 371089 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	54100e7357	Merged revisions 371061 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371061 | may | 2012-08-10 10:13:10 -0500 (Fri, 10 Aug 2012) | 10 lines Send re-register packets by GRQ (gatekeeper request) interval (close issue ASTERISK-20094) Patches: ASTERISK-20094-2.patch ........ Merged revisions 371060 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	a5800a1741	Merged revisions 371013,371022 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r371013 | rmudgett | 2012-08-09 14:11:01 -0500 (Thu, 09 Aug 2012) | 5 lines Use better libss7 detection test and move libpri compile test. ........ Merged revisions 371012 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r371022 | may | 2012-08-09 14:20:09 -0500 (Thu, 09 Aug 2012) | 10 lines Fix to resend GRQ/RRQ if RRJ (registration reject) is received (close issue ASTERISK-20094) Patches: ASTERISK-20094.patch ........ Merged revisions 371011 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371035 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	60a5d5f510	Merged revisions 370986,370989 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370986 | kmoore | 2012-08-09 12:39:52 -0500 (Thu, 09 Aug 2012) | 11 lines Correct documentation for the MeetMe x flag The documentation for the x flag for MeetMe incorrectly described its function as closing down the conference when the last marked user left. It actually causes the users with that flag to leave the conference when the last marked user exits. The functionality of this flag is not changing. ........ Merged revisions 370985 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r370989 | may | 2012-08-09 13:05:34 -0500 (Thu, 09 Aug 2012) | 5 lines change opening h323 logfile with append mode instead of overwrite ........ Merged revisions 370988 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	0e1ac09a2b	Merged revisions 370954 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370954 | elguero | 2012-08-08 17:42:05 -0500 (Wed, 08 Aug 2012) | 26 lines Fix Not Unreferencing A Spied Channel When a channel hangs up while being spied upon and the option to exit the ChanSpy application when the spied on channel hangs up is set, ast_autochan_destroy is not being called and therefore a reference to the spied upon channel is not removed. The symptom being reported was that when using func_group in the dialplan and calling "group show channels" at the cli, the spied upon channel was still being shown while "core show channels" showed that the channel was not up. This patch calls ast_autochan_destroy when a spied upon channel hangs up and the option to exit the ChanSpy application is set, removing the reference to the channel allowing the count for the group that the spied channel was part of to be decremented. (closes issue ASTERISK-17515) Reported by: Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 370952 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	9dfb7700c4	Merged revisions 370924 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370924 | kmoore | 2012-08-08 15:29:16 -0500 (Wed, 08 Aug 2012) | 9 lines Do not define a cause that doesn't actually exist AST_CAUSE_NOTDEFINED is a placeholder for usage when there is no cause information. As such, it should not be defined and translatable as a cause. ........ Merged revisions 370923 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370948 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	105b05218a	Merged revisions 370901 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370901 | rmudgett | 2012-08-08 15:04:44 -0500 (Wed, 08 Aug 2012) | 5 lines Fix the analog dial *0 flash-hook of bridged peer feature. ........ Merged revisions 370900 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370922 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	cb11c1cf4d	Merged revisions 370858 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370858 | kmoore | 2012-08-07 14:21:54 -0500 (Tue, 07 Aug 2012) | 5 lines Add missing AST_CAUSE_* -> text translations ........ Merged revisions 370856 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	64bf23b2d2	Merged revisions 370798 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370798 | mmichelson | 2012-08-06 10:02:04 -0500 (Mon, 06 Aug 2012) | 7 lines Improve debug message for temporary outbound proxies. Thanks to Paul Belanger for pointing this out. ........ Merged revisions 370797 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370817 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	adc08df2d1	Merged revisions 370772 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370772 | mmichelson | 2012-08-03 16:50:29 -0500 (Fri, 03 Aug 2012) | 41 lines Multiple revisions 370769-370771 ........ r370769 | mmichelson | 2012-08-03 16:35:00 -0500 (Fri, 03 Aug 2012) | 24 lines Fix error in the "IPorHost" section of a SIP dialstring. This is based on the review request posted by Walter Doekes (referenced lower in the commit message) The main fix here is to treat the IPorHost portion of the dial string as a temporary outbound proxy. This ensures requests get sent to the proper location. Due to the age of the request, some parts were no longer relevant. For instance, the request moved outbound proxy parsing code into a single method. This is done in a previous commit, so it was not necessary to do again. Also, the review request fixed some errors with regards to request routing for CANCEL and ACK requests. This has also been fixed in more recent commits. (closes issue ASTERISK-19677) reported by Walter Doekes Review https://reviewboard.asterisk.org/r/1859 ........ r370770 | mmichelson | 2012-08-03 16:39:35 -0500 (Fri, 03 Aug 2012) | 3 lines Remove unused variable. ........ r370771 | mmichelson | 2012-08-03 16:43:52 -0500 (Fri, 03 Aug 2012) | 5 lines Seriously? Another compilation error fixed. Somebody beat me. ........ Merged revisions 370769-370771 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	81fdb9dfc9	Merged revisions 370698 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370698 | kmoore | 2012-07-31 21:26:09 -0500 (Tue, 31 Jul 2012) | 8 lines Revert alloca changes for utils These changes were a tad overzealous in the utils directory. Unfortunately, these don't compile with a "make". ........ Merged revisions 370697 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	764c114bee	Merged revisions 370672 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370672 | mjordan | 2012-07-31 16:19:41 -0500 (Tue, 31 Jul 2012) | 24 lines Schedule pokes of registered SIP peers within a given timespan after SIP reload With a large number of SIP peers registered, performing a SIP reload causes a flood of SIP OPTIONS request packets. These are immediately sent out, and, as responses come back, can cause peers to be flagged as 'lagged' due to handling of the many response messages. This fix prevents this "packet storm" and schedules the pokes for a random time. That time varies between 1 ms and the peer's qualify time, or, if the qualify time is unknown, the global qualifyfreq setting. The committed patch has some very small modifications to the patch schmidts wrote for the review. (closes issue ASTERISK-19154) Reported by: Nicolo Mazzon patches: issue19154.patch license #6034 uploaded by schmidts Review: https://reviewboard.asterisk.org/r/1652 ........ Merged revisions 370666 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	86776cb4d5	Merged revisions 370643 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370643 | kmoore | 2012-07-31 14:57:09 -0500 (Tue, 31 Jul 2012) | 12 lines Clean up and ensure proper usage of alloca() This replaces all calls to alloca() with ast_alloca() which calls gcc's __builtin_alloca() to avoid BSD semantics and removes all NULL checks on memory allocated via ast_alloca() and ast_strdupa(). (closes issue ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/ ........ Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	26577d1d93	Merged revisions 370619 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370619 | mmichelson | 2012-07-31 10:31:57 -0500 (Tue, 31 Jul 2012) | 34 lines Help mitigate potential reinvite glare scenarios. When Asterisk servers are set up back-to-back, and direct media is to be used betweeen endpoints, it is fairly common for the two Asterisk servers to send direct media reinvites to each other simultaneously. This results in 491s and ACKs being exchanged between the servers. While the media eventually gets set up properly, the problem is that there can be a noticeable delay for the streams to stabilize. This patch adds a new directmedia option called "outgoing". With this set, an immediate direct media reinvite will only be sent if the call direction is outgoing. For incoming dialogs, an immediate direct media reinvite will not be sent, but further "reactionary" direct media reinvites may be sent. For those who are having some deja vu, that's because this patch was originally committed to trunk since there is a new configuration option added. After seeing a bug report about audio being slow to set up on SIP calls, it became apparent that this patch would be the best solution for resolving the issue. The patch is unintrusive and will have no effect unless the option is explicitly enabled. (closes issue AST-896) reported by Thomas Arimont (closes issue ASTERISK-19857) reported by Matt Jordan ........ Merged revisions 370618 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370633 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	b5a26ce601	Merged revisions 370564 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370564 | rmudgett | 2012-07-30 11:49:12 -0500 (Mon, 30 Jul 2012) | 5 lines Release B channel allocation on error path in chan_misdn. ........ Merged revisions 370563 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	ec96eda533	Merged revisions 370547 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r370547 | jrose | 2012-07-30 09:50:34 -0500 (Mon, 30 Jul 2012) | 5 lines app_meetme: Change app_meetme support level to extended from deprecated (closes issue ASTERISK-20134) Reported by: Leif Madsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	54f01be015	Merged revisions 370495 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370495 | jrose | 2012-07-25 16:12:50 -0500 (Wed, 25 Jul 2012) | 14 lines res_agi: Add message indicating need for \n character in verbose message The while loop responsible for reading AGI messages from a fastAGI service can end up looping indefinitely when an AGI script fails to indicate the end of a message with a \n character. This patch adds an indication that we are expecting a \n character to end the message to make it more clear to users that this is necessary if they are receiving this warning over and over. (issue ASTERISK-20061) Reported by: Eike Kuiper ........ Merged revisions 370494 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	e3f9e1b64d	Revert a change that broke compilation ... 1) There is no such function as ast_ref() 2) The patch was originally credited as the one uploaded by Guenther Kelleter (license 6372) via issue AST-921, but the patch committed was not the patch referenced on the issue. 3) Guenther Kelleter's patch was actually correct. It moved the ast_free above the presencechange_cleanup label. I am not committing his change as it is not technically necesary--calling ast_free(NULL) is perfectly safe and I worry that moving the ast_free outside of the label could lead to future bugs if someone ever adds another failure conditional and expects 'goto presencechange_cleanup;' to clean up after everything. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ff8e14d782	Don't attempt free of NULL ptr in pbx.c handle_presencechange ... (closes issue AST-921) Reported by: Guenther Kelleter Patches: nullptr.patch uploaded by Guenther Kelleter (license 6372) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	a2fe225529	Merged revisions 370430,370432 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370430 | kpfleming | 2012-07-24 11:54:01 -0500 (Tue, 24 Jul 2012) | 5 lines Rewrite a comment that didn't adequately explain the code it was documenting. ........ Merged revisions 370429 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r370432 | tzafrir | 2012-07-24 12:08:40 -0500 (Tue, 24 Jul 2012) | 4 lines chan_oss: fix "sample rate" error message Merged revisions 370428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	33cbee2d22	Merged revisions 370384 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370384 | kpfleming | 2012-07-23 16:09:53 -0500 (Mon, 23 Jul 2012) | 5 lines Improve documentation for the SHELL() dialplan function. ........ Merged revisions 370383 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	2dec8e2a0a	Merged revisions 370361 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370361 | kpfleming | 2012-07-23 09:51:21 -0500 (Mon, 23 Jul 2012) | 13 lines Free any datastores attached to dummy channels. Revision 370205 added the use of a datastore attached to a dummy channel to resolve a memory leak, but ast_dummy_channel_destructor() in this branch did not free datastores, resulting in a continued (but slightly smaller) memory leak. This patch backports the change to free said datastores from the Asterisk trunk. (related to issue AST-916) ........ Merged revisions 370360 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	5912f75512	Merged revisions 370271,370274,370277 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370271 | mjordan | 2012-07-19 16:37:09 -0500 (Thu, 19 Jul 2012) | 49 lines Handle extremely out of order RFC 2833 DTMF The current implementation of RFC 2833 DTMF handling in res_rtp_asterisk will, if a packet arrives out of order, drop the packet. This is to prevent duplicate ton generation in the Asterisk core. Since the RTP layer does not buffer data itself, this is the only option the RTP layer currently has for handling packets that arrive out of order. For the most part, this doesn't matter. For a particular digit, so long as a BEGIN packet arrives before the first END packet, the digit will be produced. If subsequent BEGIN packets arrive interleaved with the ENDs, they will be dropped; likewise, if the BEGIN or END packets themselves are out of order, those packets are dropped but sufficient information is conveyed to the Asterisk core to produce the appropriate digit. For certain sequences of DTMF packets - most notably when, for a particular digit, an END packet arrives before any BEGIN packet for that digit - this is a real problem. When an END arrives before any BEGINs, the END packet is dropped - but at the same time, it causes subsequent BEGIN packets for that digit to be ignored. When the next in order END packet arrives, it too is dropped - Asterisk believes that there was no initial BEGIN. The solution this patch provides is to trust the END packet to convey the information needed for the Asterisk core to produce the DTMF digit. If we receive an END packet, and it: * Has a timestamp greater then the last timestamp received from an END packet * Does not have the same sequence number as the last received sequence number (and is thus not an END packet retransmission) Then we send the END frame up to the Asterisk core. It contains enough DTMF information for Asterisk to produce the digit. On the other hand, if we receive a BEGIN or continuation packet that occurs with a timestamp equal to or less then the last END timestamp, then we've received something out of order - but we already have received enough information to produce the digit. These packets are dropped. Much thanks goes to Olle Johansson (oej) for providing the idea for this solution. Review: https://reviewboard.asterisk.org/r/2033/ (issue ASTERISK-18404) Reporter: Stephane Chazelas Tested by: Matt Jordan ........ Merged revisions 370252 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r370274 | mjordan | 2012-07-19 17:01:32 -0500 (Thu, 19 Jul 2012) | 17 lines Fix compilation error when MALLOC_DEBUG is enabled To fix a memory leak in CEL, a channel datastore was introduced whose destruction function pointer was pointed to the ast_free macro. Without MALLOC_DEBUG enabled this compiles as fine, as ast_free is defined as free. With MALLOC_DEBUG enabled, however, ast_free takes on a definition from a different place then utils.h, and became undefined. This patch resolves this by using a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this calls ast_free; when MALLOC_DEBUG is not enabled, this is defined to be ast_free, which is defined to be free. (issue AST-916) Reported by: Thomas Arimont ........ Merged revisions 370273 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r370277 | rmudgett | 2012-07-19 17:11:48 -0500 (Thu, 19 Jul 2012) | 7 lines Fix compiler warnings. gcc (GCC) 4.2.4 has problems casting away constness. ........ Merged revisions 370275 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	1440b3f1ec	Merged revisions 370206 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370206 | kpfleming | 2012-07-18 14:14:09 -0500 (Wed, 18 Jul 2012) | 19 lines Resolve severe memory leak in CEL logging modules. A customer reported a significant memory leak using Asterisk 1.8. They have tracked it down to ast_cel_fabricate_channel_from_event() in main/cel.c, which is called by both in-tree CEL logging modules (cel_custom.c and cel_sqlite3_custom.c) for each and every CEL event that they log. The cause was an incorrect assumption about how data attached to an ast_channel would be handled when the channel is destroyed; the data is now stored in a datastore attached to the channel, which is destroyed along with the channel at the proper time. (closes issue AST-916) Review: https://reviewboard.asterisk.org/r/2053/ ........ Merged revisions 370205 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago