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Author	SHA1	Message	Date
Automerge script	8d9bfd3342	Merged revisions 370184 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370184 | kpfleming | 2012-07-18 12:13:07 -0500 (Wed, 18 Jul 2012) | 10 lines Ensure that all ast_datastore_info structures are 'const'. While addressing a bug, I came across a instance of 'struct ast_datastore_info' that was not declared 'const'. Since the API already expects them to be 'const', this patch changes the declarations of all existing instances that were not already declared that way. ........ Merged revisions 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370202 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	b2ab0e1d0c	Merged revisions 370132 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370132 | wdoekes | 2012-07-16 14:52:45 -0500 (Mon, 16 Jul 2012) | 11 lines Code cleanup and bugfix in chan_sip outboundproxy parsing. The bug was clearing the global outboundproxy when a peer-specific outboundproxy was bad. The cleanup reduces duplicate code. Review: https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark Michelson ........ Merged revisions 370131 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	a0fec72b8b	Merged revisions 370082 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370082 | kmoore | 2012-07-16 08:51:57 -0500 (Mon, 16 Jul 2012) | 8 lines Add comments about the BUILD_NATIVE change This is a significant change and mention of it should have gone into UPGRADE.txt and CHANGES. ........ Merged revisions 370081 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	2190019e49	Merged revisions 370015,370025 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r370015 | kmoore | 2012-07-12 15:05:45 -0500 (Thu, 12 Jul 2012) | 11 lines Include Expires header for SIP PUBLISH requests RFC3903 requres SIP PUBLISH requests to have Expires headers, so add them. Review: https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth ........ Merged revisions 370014 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r370025 | rmudgett | 2012-07-12 15:20:02 -0500 (Thu, 12 Jul 2012) | 8 lines Add missing ast_hangup() calls on some analog exception paths. Make starting analog_ss_thread() or __analog_ss_thread() failure paths hangup the channel. ........ Merged revisions 370017 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	825c3eaf2b	Merged revisions 369994 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369994 | kmoore | 2012-07-12 13:55:17 -0500 (Thu, 12 Jul 2012) | 12 lines Prevent double uri_escaping in chan_sip when pedantic is enabled If pedantic mode is enabled, outbound invites will have double-escaped contacts. This avoids setting an already-escaped string into a field where it is expected to be unescaped. (closes issue ASTERISK-20023) Reported-by: Walter Doekes ........ Merged revisions 369993 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370013 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	2e1130db73	Merged revisions 369971 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369971 | elguero | 2012-07-12 09:25:45 -0500 (Thu, 12 Jul 2012) | 14 lines Correct Documentation For DEC Function The documentation for DEC in func_math.c was incorrect. Looks like a copy and paste error. (Closes issue ASTERISK-20095) Reported by: Billy Chia Tested by: Michael L. Young Patches: func_math.patch uploaded by Billy Chia (license 6381) ........ Merged revisions 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f59ef2b922	Merged revisions 369938 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369938 | tilghman | 2012-07-11 12:12:28 -0500 (Wed, 11 Jul 2012) | 11 lines Allow the REALTIME() function to report errors back to the caller. Also, do more error checking on the arguments specified to the REALTIME() function and clarify the documentation. While I was editing the file, a few coding guidelines fixups, as well. Review: https://reviewboard.asterisk.org/r/2031/ ........ Merged revisions 369937 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369958 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	09a61e14d1	Merged revisions 369871 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369871 | kmoore | 2012-07-10 08:35:30 -0500 (Tue, 10 Jul 2012) | 12 lines Improve Goto and GotoIf related documentation Correct documentation on labeliftrue and labeliffalse parameters of GotoIf() and update several other locations that use the same syntax. (closes issue ASTERISK-20007) Patch-by: Leif Madsen Reported-by: WIMPy ........ Merged revisions 369869 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369889 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Joshua Colp	890f7bf9e7	Add support for exposing the received contact URI and also for setting the request URI in messages. ... (closes issue AST-911) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	ead4555a90	Merged revisions 369819 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369819 | qwell | 2012-07-09 12:06:40 -0500 (Mon, 09 Jul 2012) | 9 lines Add Digium phones context to sip_notify sample config. This makes it so that they can be reconfigured remotely. (closes issue ASTERISK-19910) ........ Merged revisions 369818 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369836 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f6c99d233e	Merged revisions 369793 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369793 | jrose | 2012-07-09 09:43:49 -0500 (Mon, 09 Jul 2012) | 9 lines chan_sip: Fix small behavioral change accidentally introduced in r369750 When removing the warning for AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed the return value, which would likely make the indication not be sent in audio. This fixes that while still removing the warning message. ........ Merged revisions 369792 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	d8930ab6d1	Merged revisions 369751 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369751 | jrose | 2012-07-06 16:02:37 -0500 (Fri, 06 Jul 2012) | 12 lines chan_sip: Add case for FLASH control frames so that we don't display a warning. chan_sip channels can receive flash control frames when connected to analog phones and possibly for other reasons. There really isn't a reason to warn when these frames are received, we can safely ignore them. Patches: dahdi_sip_flash.diff uploaded by Jonathan Rose (license 6182) ........ Merged revisions 369750 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369763 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	19afbc2af2	Merged revisions 369732 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369732 | mmichelson | 2012-07-06 13:47:05 -0500 (Fri, 06 Jul 2012) | 21 lines Remove a superfluous and dangerous freeing of an SSL_CTX. The problem here is that multiple server sessions share a SSL_CTX. When one session ended, the SSL_CTX would be freed and set NULL, leaving the other sessions unable to function. The code being removed is superfluous because the SSL_CTX structures for servers will be properly freed when ast_ssl_teardown is called. (closes issue ASTERISK-20074) Reported by Trevor Helmsley Patches: ASTERISK-20074.diff uploaded by Mark Michelson (license #5049) Testers: Trevor Helmsley ........ Merged revisions 369731 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	b4dcb67e17	Merged revisions 369709 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369709 | mmichelson | 2012-07-06 10:23:28 -0500 (Fri, 06 Jul 2012) | 14 lines Fix bridging thread leak. The bridge thread was exiting but was never being reaped using pthread_join(). This has been fixed now by calling pthread_join() in ast_bridge_destroy(). (closes issue ASTERISK-19834) Reported by Marcus Hunger Review: https://reviewboard.asterisk.org/r/2012 ........ Merged revisions 369708 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369725 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f4c2e137bb	Merged revisions 369653 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369653 | kmoore | 2012-07-05 14:12:33 -0500 (Thu, 05 Jul 2012) | 20 lines Resolve heap corruption issue with voicemail The heard and deleted arrays in the voicemail state structure were not handled properly following the memory leak fix in r354890 and a fix for an invalid free in r356797. This could result in accessing and writing into freed memory. The allocation for these arrays has been reworked to avoid the possibility of invalid frees, access of freed memory, and crashes that were occurring as a result of this. Locking around accesses and modifications of the voicemail state structure members dh_arraysize, heard, and deleted has been added to prevent simultaneous modification and access when IMAP storage is in use. If IMAP storage is not in use, this locking is not compiled in. Review: https://reviewboard.asterisk.org/r/1994/ (closes issue ASTERISK-19923) ........ Merged revisions 369652 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	1bf80b527c	Merged revisions 369627 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369627 | mjordan | 2012-07-05 12:02:53 -0500 (Thu, 05 Jul 2012) | 18 lines Do not send a BYE when a provisional response arrives during a re-INVITE Commits r369557 and r369579 were done to improve handling of re-INVITEs when the UA that was supposed to receive the re-INVITE fails to respond. A limitation of those patches occurred when a UA sent a provisional response to the re-INVITE. This triggered a sending of a BYE in check_pending. This patch tweaks the handling of the re-INVITE such that a BYE is not sent in response to those messages. (issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies patches: (reinvite_tweak.diff license #5012 by Steve Davies) ........ Merged revisions 369626 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369643 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f477a9e831	Merged revisions 369580 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369580 | twilson | 2012-07-03 12:02:18 -0500 (Tue, 03 Jul 2012) | 11 lines More improvements to re-INVITEs timing out after a provisional response There is no need to call check_pendings() on a final response to an INVITE when destroying the scheduler entry as it will be done later during normal processing. (issue ASTERISK-19992) ........ Merged revisions 369579 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	ad64385217	Merged revisions 369558 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369558 | twilson | 2012-07-03 09:34:22 -0500 (Tue, 03 Jul 2012) | 14 lines Better handle re-INVITEs with provisional but no final repsonses A previous attempt at fixing this issue had negative side effects related to attended transfers which this patch should resolve. Many thanks to Steve Davies for all of the good suggestions and testing. (closes issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies, Terry Wilson Review: https://reviewboard.asterisk.org/r/2009/ ........ Merged revisions 369557 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	4fb96676bd	Merged revisions 369511 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r369511 | mmichelson | 2012-06-29 15:28:10 -0500 (Fri, 29 Jun 2012) | 3 lines Fix apparent copy and paste error where incorrect "glue" is used. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	c2711a357e	Merged revisions 369491 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369491 | file | 2012-06-29 11:54:11 -0500 (Fri, 29 Jun 2012) | 5 lines With some configurations a transport is not actually specified so assume UDP in these cases. ........ Merged revisions 369490 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f51af261fe	Merged revisions 369472 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369472 | file | 2012-06-29 10:30:47 -0500 (Fri, 29 Jun 2012) | 10 lines Make the address family filter specific to the transport. (closes issue ASTERISK-16618) Reported by: Leif Madsen Review: https://reviewboard.asterisk.org/r/1667/ ........ Merged revisions 369471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	23db01bcc8	Merged revisions 369437 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369437 | twilson | 2012-06-27 16:10:01 -0500 (Wed, 27 Jun 2012) | 16 lines Clean up after a reinvite that never gets a final response The basic problem is that if a re-INVITE is sent by Asterisk and it receives a provisional response, but no final response, then the dialog is never torn down. In addition to leaking memory, this also leaks file descriptors and will eventually lead to Asterisk no longer being able to process calls. This patch just keeps track of whether there is an outstanding re-INVITE, and if there is goes ahead and cleans up everything as though there was no outstanding reinvite. (closes issue ASTERISK-19992) ........ Merged revisions 369436 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	3fe87f3c47	Merged revisions 369391 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369391 | mjordan | 2012-06-26 08:22:42 -0500 (Tue, 26 Jun 2012) | 15 lines Fix crash in unloading of res_adsi module When res_adsi is unloaded, it removes the ADSI functions that it previously installed by passing a NULL adsi_funcs pointer to ast_adsi_install_funcs. This function was not checking whether or not the adsi_funcs pointer passed in was NULL before dereferencing it to check whether or not the version of the functions matches what the core was expecting it. This patch makes it so that the version is only checked if a potentially valid adsi_funcs pointer was passed in. Passing in NULL removes the installed functions, bypassing the version check. ........ Merged revisions 369390 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	4478e3580b	Merged revisions 369369 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369369 | mjordan | 2012-06-25 14:36:02 -0500 (Mon, 25 Jun 2012) | 29 lines Fix incorrect duration reporting in CDRs created in batch mode Certain places in core/cdr.c would, if the duration value were 0, calculate the duration as being the delta between the current time and the time at which the CDR record was started. While this does not typically cause a problem in non-batch mode, this can cause an issue in batch mode where CDR records are gathered and written long after those calls have ended. In particular, this affects calls that were never answered, as those are expected to have a duration of 0. Often, this would result in CDR logs with a significant number of calls with lengthy durations, but dispositions of "BUSY". Note that this does not affect cdr_csv, as that backend does not use ast_cdr_getvar and instead directly reports the duration value. The affected core backends include cdr_apative_odbc and cdr_custom; other extended or deprecated CDR backends may potentially still directly manipulate the duration values. (issue ASTERISK-19860) Reported by: Thomas Arimont (issue AST-883) Reported by: Thomas Arimont Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1996/ ........ Merged revisions 369351 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	b25a80690c	Merged revisions 369353 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369353 | mmichelson | 2012-06-25 14:16:52 -0500 (Mon, 25 Jun 2012) | 14 lines Re-fix how local tag is generated when sending a 481 to an INVITE. Match our local tag to whatever to-tag was sent in the initial INVITE. Because the size of the to-tag may not fit in the buffer in the sip_pvt, it has been changed to a string field. (closes issue ASTERISK-19892) reported by Walter Doekes Review: https://reviewboard.asterisk.org/r/1977 ........ Merged revisions 369352 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	108f718fd5	Merged revisions 369325,369328 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369325 | mmichelson | 2012-06-25 10:52:42 -0500 (Mon, 25 Jun 2012) | 20 lines Multiple revisions 369323-369324 ........ r369323 | mmichelson | 2012-06-25 10:35:43 -0500 (Mon, 25 Jun 2012) | 9 lines Eliminate embedding of res_adsi.so module. The way this is done is to stop using the optional API. Instead, res_adsi.so, when loaded fills in a table of function pointers. Review: https://reviewboard.asterisk.org/r/1991 ........ r369324 | mmichelson | 2012-06-25 10:50:17 -0500 (Mon, 25 Jun 2012) | 2 lines Forgot to svn add this file in my last commit. ........ Merged revisions 369323-369324 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r369328 | rmudgett | 2012-06-25 10:59:28 -0500 (Mon, 25 Jun 2012) | 15 lines Fix Bridge application occasionally returning to the wrong location. * Fix do_bridge_masquerade() getting the resume location from the zombie channel. The code must not touch a clone channel after it has masqueraded it. The clone channel has become a zombie and is starting to hangup. (closes issue ASTERISK-19985) Reported by: jamicque Patches: jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: jamicque ........ Merged revisions 369327 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	03b52a9b84	Merged revisions 369303 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369303 | mmichelson | 2012-06-25 09:23:16 -0500 (Mon, 25 Jun 2012) | 14 lines Be more consistent with the return code for requests received from invalid domain. When Asterisk receives an INVITE from an external domain when allowexternaldomains=no send a 403 instead of a 404. This is consistent with Asterisk's behavior when receiving a REGISTER in this situation. (Closes issue ASTERISK-19601) Reported by Matthew Jordan Patches: ASTERISK-19601-no401.patch uploaded by Mark Michelson (License #5049) ........ Merged revisions 369302 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369322 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	17a5e814f5	Merged revisions 369283 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369283 | rmudgett | 2012-06-22 19:12:27 -0500 (Fri, 22 Jun 2012) | 22 lines Fix Bridge application and AMI Bridge action error handling. * Fix AMI Bridge action disconnecting the AMI link on error. * Fix AMI Bridge action and Bridge application not checking if their masquerades were successful. * Fix Bridge application running the h-exten when it should not. * Made do_bridge_masquerade() return if the masquerade was successful so the Bridge application and AMI Bridge action could deal with it correctly. * Made bridge_call_thread_launch() hangup the passed in channels if the bridge_call_thread fails to start. Those channels would have been orphaned. * Made builtin_atxfer() check the success of the transfer masquerade setup. ........ Merged revisions 369282 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f3cc355870	Merged revisions 369259,369263 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369259 | rmudgett | 2012-06-22 16:37:05 -0500 (Fri, 22 Jun 2012) | 5 lines Check if PBX was started and fix F and F(x) action logic in Dial application. ........ Merged revisions 369258 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r369263 | rmudgett | 2012-06-22 17:09:29 -0500 (Fri, 22 Jun 2012) | 5 lines Explicitly check caller hangup in app Queue rather than a polluted res2 value. ........ Merged revisions 369262 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	9c288ea612	Merged revisions 369236,369239 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369236 | rmudgett | 2012-06-22 15:49:33 -0500 (Fri, 22 Jun 2012) | 5 lines Change incorrect chan_sip zombie hangup debug message. They are all zombies now. ........ Merged revisions 369235 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r369239 | rmudgett | 2012-06-22 16:04:25 -0500 (Fri, 22 Jun 2012) | 5 lines Check if PBX was started for generic CCSS recall. ........ Merged revisions 369238 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	41de4b94a1	Merged revisions 369215 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369215 | twilson | 2012-06-22 14:34:59 -0500 (Fri, 22 Jun 2012) | 9 lines Don't crash on a guest directmedia call A sip_pvt may not have relatedpeer set if a call doesn't match up with a peer. If there is no relatedpeer, there is no direct media ACL to apply, so just return that it is allowed. ........ Merged revisions 369214 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	1744bfd2e4	Merged revisions 369206 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369206 | kmoore | 2012-06-22 12:23:26 -0500 (Fri, 22 Jun 2012) | 11 lines Don't parse media stream state for SIP video streams The sendonly/recvonly/sendrecv/inactive media stream attributes were parsed for video, but nothing was ever done with them. With this code removed, an UNSUPPORTED message is produced when these attributes are used in conjunction with a video stream which is the better behavior since they were never really supported in the first place. ........ Merged revisions 369195 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	46f6c80878	Merged revisions 369147 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369147 | may | 2012-06-20 12:36:27 -0500 (Wed, 20 Jun 2012) | 10 lines fix locking issue on empty callList (issue ASTERISK-19298) Reported by: Dmitry Melekhov Patches: ASTERISK-18322-2.patch ........ Merged revisions 369146 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	81f478ae7c	Merged revisions 369109 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369109 | elguero | 2012-06-19 21:04:58 -0500 (Tue, 19 Jun 2012) | 23 lines Fix NULL pointer segfault in ast_sockaddr_parse() While working with ast_parse_arg() to perform a validity check, a segfault occurred. The segfault occurred due to passing a NULL pointer to ast_sockaddr_parse() from ast_parse_arg(). According to the documentation in config.h, "result pointer to the result. NULL is valid here, and can be used to perform only the validity checks." This patch fixes the segfault by checking for a NULL pointer. This patch also adds documentation to netsock2.h about why it is necessary to check for a NULL pointer. (Closes issue ASTERISK-20006) Reported by: Michael L. Young Tested by: Michael L. Young Patches: asterisk-20006-netsock-null-ptr.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1990/ ........ Merged revisions 369108 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369125 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	a67c5aa48e	Merged revisions 369091 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r369091 | may | 2012-06-19 18:32:06 -0500 (Tue, 19 Jun 2012) | 9 lines check rtptimeouts in ooh323 channels as per config file (rtp voice, video, udptl except rtcp) (closes issue ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches: 19179-ooh323-ast10.patch ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369107 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	9c05926dca	Merged revisions 369067 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369067 | mmichelson | 2012-06-19 10:37:37 -0500 (Tue, 19 Jun 2012) | 17 lines Fix request routing issue when outboundproxy is used. Asterisk was incorrectly setting the destination of CANCELs and ACKs for error responses to the URI of the initial INVITE. This resulted in further requests, such as INVITEs with authentication credentials, to be routed incorrectly. Instead, when these CANCEL or ACKs are to be sent, we should simply keep the destination the same as what it previously was. There is no need to alter it any. (closes issue ASTERISK-20008) Reported by Marcus Hunger Patches: ASTERISK-20008.patch uploaded by Mark Michelson (license #5049) ........ Merged revisions 369066 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	9cd0f2c7f6	Merged revisions 369044 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369044 | rmudgett | 2012-06-18 13:11:30 -0500 (Mon, 18 Jun 2012) | 12 lines Fix monitoring calls put in a parking lot. * Fix a regression that was introduced by -r366167 which effectively disabled monitoring parked calls. (closes issue ASTERISK-20012) Reported by: sdolloff Tested by: rmudgett ........ Merged revisions 369043 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369056 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	96a5878bd6	Fix voicemail API tests by using the correct argument order for create/destroy. ... ........ Merged revisions 369024 from http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369026 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	8aef9fdedd	Merged revisions 369005 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r369005 | kpfleming | 2012-06-15 11:07:08 -0500 (Fri, 15 Jun 2012) | 22 lines Multiple revisions 369001-369002 ........ r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines Add support-level indications to many more source files. Since we now have tools that scan through the source tree looking for files with specific support levels, we need to ensure that every file that is a component of a 'core' or 'extended' module (or the main Asterisk binary) is explicitly marked with its support level. This patch adds support-level indications to many more source files in tree, but avoids adding them to third-party libraries that are included in the tree and to source files that don't end up involved in Asterisk itself. ........ r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines Add a script to enable finding source files without support-levels defined. ........ Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@369023 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	93902647dc	Remove some symbol exports that got missed in the removal of global symbols. ... (issue AST-807) (issue AST-901) (issue AST-908) ........ Merged revisions 368998 from http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@368999 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	198b84de58	These functions that were moved need to be static. ... Also wrap test functions in a #ifdef. (issue AST-807) (issue AST-901) (issue AST-908) ........ Merged revisions 368964 from http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@368965 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	6b9dfb9061	Remove global symbol requirement from app_voicemail. ... This uses the existing "function installation" stuff that already existed for other functions, like getting message counts. (closes issue AST-807) (issue AST-901) (issue AST-908) Review: https://reviewboard.asterisk.org/r/1965/ ........ Merged revisions 368962 from http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@368963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	e58c3f3441	Merged revisions 368947 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r368947 | mjordan | 2012-06-14 12:31:33 -0500 (Thu, 14 Jun 2012) | 21 lines AST-2012-009: Fix crash in chan_skinny due to Key Pad Button Message handling AST-2012-008 (r367844) fixed a denial of service attack exploitable in the Skinny channel driver that occurred when certain messages are sent after a previously registered station sends an Off Hook message. Unresolved in that patch is an issue in the Asterisk 10 releases, wherein, if a Station Key Pad Button Message is processed after an Off Hook message, the channel driver will inappropriately dereference a NULL pointer. This patch fixes those places where the message handling or the channel callback functions would attempt to dereference the line's pointer to the device. (issue ASTERISK-19905) Reported by: Christoph Hebeisen Tested by: mjordan, Christoph Hebeisen Patches: AST-2012-009-10.diff uploaded by mjordan (license 6283) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@368960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	1a4c7f29f3	Merged revisions 368928 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r368928 | mmichelson | 2012-06-14 10:25:23 -0500 (Thu, 14 Jun 2012) | 10 lines Revert Makefile change to remove embedding res_adsi.so The change has resulted in a linking error for certain versions of GCC. This is much worse than the original issue, so for now, temporarily revert the change. A more thorough change will be sought out. ........ Merged revisions 368927 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@368946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	0c709ed085	Merged revisions 368895,368899 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r368895 | mjordan | 2012-06-13 15:27:28 -0500 (Wed, 13 Jun 2012) | 21 lines Mark res_smdi/res_adsi as 'core' supported modules Recently, various issues surrounding weak attributes have caused problems with modules that rely on that feature to be enabled in menuselect. This includes app_voicemail and chan_dahdi, as they both rely upon res_smdi and res_adsi, which, in certain circumstances, may not be enabled by default in menuselect. Because res_smdi/res_adsi are dependencies for chan_dahdi/app_voicemail, this patch marks both as 'core' supported modules. This will allow both app_voicemail and chan_dahdi to be enabled as well, regardless of whether or not that system supports weak attributes. (issue AST-900) Reported by: Thomas Arimont (issue AST-885) Reported by: Denis Alberto Martinez ........ Merged revisions 368894 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r368899 | mmichelson | 2012-06-13 16:13:30 -0500 (Wed, 13 Jun 2012) | 19 lines Fix a deadlock that occurs when func_volume is used on a local channel. This was discovered by trying to perform a call forward to an extension that makes use of func_volume. When the local channel is optimized away, the datastore on the local;2 channel would have its audiohook destroyed rather than detaching the audiohook from the channel and then destroying it. With this patch, func_volume's datastore destructor takes the proper route of detaching the audiohook and then destroying it. (closes issue ASTERISK-19611) reported by Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark Michelson (license #5049) ........ Merged revisions 368898 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@368917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	64aeb84139	Merged revisions 368885 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r368885 | mmichelson | 2012-06-13 14:36:39 -0500 (Wed, 13 Jun 2012) | 11 lines Remove forced linking of res_adsi.o In GCC 4.5+ the result is that Asterisk has a phantom module loaded at startup, claiming to be res_adsi. (closes issue ASTERISK-19920) reported by Leif Madsen ........ Merged revisions 368873 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@368893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	18527968d2	Merged revisions 368831,368853 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r368831 | mjordan | 2012-06-12 13:30:06 -0500 (Tue, 12 Jun 2012) | 24 lines Do not perform install on existing directories If a directory already exists, performing a 'make install' will remove the permissions associated with the current directory and replace them with the permissions of the user executing the install. This patch changes this behavior to only perform an install on the directory if the directory does not exist. Thus, if a user later changes the permissions on that directory, those permissions will be preserved in subsequent installs. Review: https://reviewboard.asterisk.org/r/1986 Review: https://reviewboard.asterisk.org/r/1864 (closes issue ASTERISK-19492) Reported by: Karl Fife Tested by: Paul Belanger, Tilghman Lesher patches: ASTERISK-19492 by pabelanger (uploaded by mjordan) ........ Merged revisions 368830 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r368853 | mjordan | 2012-06-13 09:30:34 -0500 (Wed, 13 Jun 2012) | 11 lines Do not install empty directories; add ASTLIBDIR r368830 modified the installation script to only create a directory if that directory does not exist. If some directory variable was empty, it would attempt to create the empty location. It also failed to create the ASTLIBDIR directory. This patch fixes it such that the correct directories are made and only created if a value specifying them actually exists. ........ Merged revisions 368852 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@368872 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	8bef8ed1e3	Let's fix the 1.8-merged prop, to give automerge the best chance at succeeding. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@368826 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	cfef0ba38d	Multiple revisions 368721,368739,368760,368808 ... ........ r368721 | kmoore | 2012-06-11 09:11:14 -0500 (Mon, 11 Jun 2012) | 8 lines Fix compilation in dev-mode Backport a compilation fix in md5.c from trunk that only showed up in dev-mode under certain compiler versions. ........ Merged revisions 368719 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r368739 | kmoore | 2012-06-11 10:15:07 -0500 (Mon, 11 Jun 2012) | 10 lines Fix coverity UNUSED_VALUE findings in core support level files Most of these were just saving returned values without using them and in some cases the variable being saved to could be removed as well. (issue ASTERISK-19672) ........ Merged revisions 368738 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r368760 | rmudgett | 2012-06-11 12:08:50 -0500 (Mon, 11 Jun 2012) | 17 lines Fix deadlock potential with ast_set_hangupsource() calls. Calling ast_set_hangupsource() with the channel lock held can result in a deadlock because the function also locks the bridged channel. (issue ASTERISK-19537) (closes issue AST-891) Reported by: Guenther Kelleter Tested by: Guenther Kelleter (closes issue ASTERISK-19801) Reported by: Alec Davis ........ Merged revisions 368759 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ r368808 | mmichelson | 2012-06-12 10:37:38 -0500 (Tue, 12 Jun 2012) | 15 lines Set the Caller ID "tag" on peers even if remote party information is present. On incoming calls, we were setting the cid_tag on the dialog only if there was no remote party information (Remote-Party-ID or P-Asserted-Identity) present. The Caller ID tag is an invented parameter, though, and should be set no matter the circumstance. (closes issue ASTERISK-19859) Reported by Thomas Arimont (closes issue AST-884) Reported by Trey Blancher ........ Merged revisions 368807 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368721,368739,368760,368808 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@368823 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	717f248612	Let's try using an automerge-propname, since we have multiple heads. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@368811 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago