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Author	SHA1	Message	Date
Automerge script	f31dd4fb9b	Merged revisions 375026 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r375026 | mmichelson | 2012-10-15 16:06:42 -0500 (Mon, 15 Oct 2012) | 22 lines Fix some potential misuses of ast_str in the code. Passing an ast_str pointer by value that then calls ast_str_set(), ast_str_set_va(), ast_str_append(), or ast_str_append_va() can result in the pointer originally passed by value being invalidated if the ast_str had to be reallocated. This fixes places in the code that do this. Only the example in ccss.c could result in pointer invalidation though since the other cases use a stack-allocated ast_str and cannot be reallocated. I've also updated the doxygen in strings.h to include notes about potential misuse of the functions mentioned previously. Review: https://reviewboard.asterisk.org/r/2161 ........ Merged revisions 375025 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@375043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	b69daa02a4	Merged revisions 374991 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374991 | tzafrir | 2012-10-14 04:40:24 -0500 (Sun, 14 Oct 2012) | 12 lines Update config.guess and config.sub: 2012-10-10 Update config.guess and config.sub to revision fb456b34ef4aa02b95dc6be69aaa66fa94a844fb from the savannah.gnu.org git repo. Adds support for e.g. aarch64 (ARM 64bit). config.guess:timestamp='2012-09-25' config.sub:timestamp='2012-10-10' ........ Merged revisions 374977 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374994 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	65d82246a9	Avoid a segfault on invalid format names ... If a format name was not found by ast_getformatbyname, a NULL pointer would be passed into ast_format_rate and immediately dereferenced. This ensures that a valid pointer is used since the structure is already allocated on the stack. (closes issue DPH-523) Reported-by: Steve Pitts git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	6f9104870e	Merged revisions 374906 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374906 | mmichelson | 2012-10-12 11:11:30 -0500 (Fri, 12 Oct 2012) | 28 lines Do not use a FILE handle when doing SIP TCP reads. This is used to solve an issue where a poll on a file descriptor does not necessarily correspond to the readiness of a FILE handle to be read. This change makes it so that for TCP connections, we do a recv() on the file descriptor instead. Because TCP does not guarantee that an entire message or even just one single message will arrive during a read, a loop has been introduced to ensure that we only attempt to handle a single message at a time. The tcptls_session_instance structure has also had an overflow buffer added to it so that if more than one TCP message arrives in one go, there is a place to throw the excess. Huge thanks goes out to Walter Doekes for doing extensive review on this change and finding edge cases where code could fail. (closes issue ASTERISK-20212) reported by Phil Ciccone Review: https://reviewboard.asterisk.org/r/2123 ........ Merged revisions 374905 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374923 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	9a99f40468	Merged revisions 374844 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374844 | mjordan | 2012-10-11 10:43:19 -0500 (Thu, 11 Oct 2012) | 29 lines Fix incorrect billing duration reported when batch mode is enabled Similar to r369351, the billing duration can be skewed when batch mode is enabled. This happened much more rarely than the duration, as it only occured when the call was answered (thereby indicating an actual answer time) and immediately hung up on (indicating a billsec of 0). Since a billing time of '0' can either mean that the call immediately ended or that the CDR was improperly answered, we have to use additional information to know whether or not we can trust the CDR billsec value. Prior to this patch, we looked to see if we had a valid answer time. If we did, and billsec was zero, we used the current time to calculate what billsec value we could from the CDR being written. If batch mode is enabled, this will incorrectly report a billsec value being much greater than the actual duration of the call. Instead of relying on the presence of an answer time to know whether or not we can re-calculate the billsec for the CDR, we now also use the presence of the CDR's end time to know if we need to re-calculate or whether we can trust the billsec value that we have. This prevents erroneous jumps in the billsec value, while still making sure that in the worst case, some billing time will be calculated. (closes issue AST-1016) Reported by: Thomas Arimont Tested by: Thomas Arimont ........ Merged revisions 374843 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374876 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	3b57018627	Merged revisions 374803 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374803 | rmudgett | 2012-10-10 15:55:44 -0500 (Wed, 10 Oct 2012) | 30 lines app_queue: Made pass connected line updates from the caller to ringing queue members. Party A calls Party B Party B puts Party A on hold. Party B calls a queue. Ringing queue member D sees Party B identification. Party B transfers Party A to the queue. Queue member D does not get a connected line update for Party A. Queue member D answers the call and still sees Party B information. However, if Party A later transfers the call to Party C then queue member D gets a connected line update for Party C. * Made pass connected line updates from the caller to queue members while the queue members are ringing. (closes issue AST-1017) Reported by: Thomas Arimont (closes issue ABE-2886) Reported by: Thomas Arimont Tested by: rmudgett ........ Merged revisions 374801 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 374802 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	4aa17120d3	Merged revisions 374763 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374763 | rmudgett | 2012-10-09 17:19:26 -0500 (Tue, 09 Oct 2012) | 15 lines Fix execution of 'i' extension due to uninitialized variable. The fix for ASTERISK-18243 added code that could potentially use dst_exten[] uninitialized. As a result the 'i' exten may not be executed when it should. (closes issue ASTERISK-20455) Reported by: Richard Miller Patches: pbx-1.8.16.0.diff (license #5685) patch uploaded by Richard Miller Made some cosmetic modifications. ........ Merged revisions 374758 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	711d68654a	Merged revisions 374728 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374728 | rmudgett | 2012-10-08 17:29:47 -0500 (Mon, 08 Oct 2012) | 15 lines dahdi.conf.sample: Add description for "buffers" setting. This contains an edited version of the patch originally uploaded by John Bigelow. (closes issue ASTERISK-14435) Reported by: John Bigelow Patches: buffers.patch (license #5091) patch uploaded by John Bigelow 0001-dahdi.conf.sample-Add-description-for-buffers-settin.patch (license #5417) patch uploaded by Shaun Ruffell Modified ........ Merged revisions 374727 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374753 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	04c3261685	Merged revisions 374695 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374695 | rmudgett | 2012-10-08 16:11:41 -0500 (Mon, 08 Oct 2012) | 34 lines Fix deletion of unopenable spool files. If scan_service() cannot open the spool file, it logs a message saying that it will delete the file and calls remove_from_queue() to do it. However, remove_from_queue() fails to delete the spool file because struct outgoing has not yet been fully initialized. * Merged allocating a new struct outgoing and init_outgoing() into new_outgoing(). Allocation is initialization. * Made apply_outgoing() not initialize the spool filename in struct outgoing. * Made apply_outgoing() call ast_trim_blanks() and ast_skip_blanks() rather than manually inlining them. * Reduced indentation levels in apply_outgoing(). * Fixed a garbled comment in remove_from_queue(). * Reworked scan_service() to simplify it. (closes issue ASTERISK-17231) Reported by: David Chappell Patches: spool_open_failure.diff (license #4997) patch uploaded by David Chappell Started with this patch. ........ Merged revisions 374686 from http://svn.asterisk.org/svn/asterisk/branches/1.8 * Fixed some memory leaks on of nominal paths in init_outgoing() when merging into the new_outgoing() function dealing with o->capabilities. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	1f39d3738f	Merged revisions 374652 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r374652 | mjordan | 2012-10-08 08:46:27 -0500 (Mon, 08 Oct 2012) | 46 lines Resolve issues in ConfBridge regarding marked, waitmarked, and unmarked users Thank's to Neil Tallim (flan)'s tireless testing, issue reporting, and patches it became clear that app_confbridge had some complex logic in how it handled interactions between marked, waitmarked, and unmarked users. In particular, there were some areas in which the interactions between the users resulted in inconsistent behavior, and app_confbridge was missing logic in how to handle some corner cases. Some areas included: * Poor handling of mixing unmarked and waitmarked users * Inconsistencies in how MOH and muting was applied to various users * Handling of various announcements for different user profile options flan's patches seem to fix the various issues, but highlighted how hard the code could be to maintain. In an attempt to make things easier to maintain and to more fully enumerate the various cases that exist, this patch breaks up the logic into a state machine-like setup. Please note that the various state transitioned are documented on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Confbridge+state+changes Review: //https://reviewboard.asterisk.org/r/2072/ Note that for the following issues, mjordan uploaded the patch, although it was written by twilson. Any contributor license discrepency is due to that. (closes issue ASTERISK-19562) Reported by: flan Tested by: flan, mjordan, jrose patches: bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283) (closes issue ASTERISK-19726) Reported by: flan Tested by: flan patches: bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283) (closes issue ASTERISK-20181) Reported by: Jonathan White Tested by: Jonathan White patches: bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	a74a6a2794	Merged revisions 374586 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374586 | dlee | 2012-10-05 15:23:14 -0500 (Fri, 05 Oct 2012) | 34 lines Multiple revisions 374570,374581 ........ r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) | 22 lines Improve AMI long line error handling In AMI's parser, when it receives a long line (> 1024 characters), it discards that line, but continues to process the message normally. Typically, this is not a problem because a) who has lines that long and b) usually a discarded line results in an invalid message. But if that line is specifying an optional field, then the message will be processed, you get a 'Response: Success', but things don't work the way you expected them to. This patch changes the behavior when a line-too-long parse error occurs. * Changes the log message to avoid way-too-long (and truncated anyways) log messages * Adds a 'parsing' status flag to Response: Success * Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line is too long * Responds with an appropriate error if parsing != MESSAGE_OKAY (closes issue AST-961) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2142/ ........ r374581 | dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line I've committed too much. Reverting part of r374570. ........ Merged revisions 374570,374581 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374610 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	65b072b64d	Merged revisions 374537 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374537 | rmudgett | 2012-10-05 13:25:20 -0500 (Fri, 05 Oct 2012) | 162 lines Merged revisions 374515-374535 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500 (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode * Made setup_bc() static. Patches: patch1_unused-code.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan states Patches: patch2_unused-states.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012) | 16 lines chan_misdn: Remove unnecessary null pointer checks and checks for stack->nt * cleanup_bc() is always called with valid bc (or it would've crashed before). * Value of stack->nt is known in advance at some places. * Rename handle_event() to handle_event_te(), handle_frm() to handle_frm_te(). Patches: patch3_checks.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 ................ r374518 | rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Fix spelling in log messages Patches: patch4_spelling.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374519 | rmudgett | 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is emptied, cleaned and set not in use, although misdn_lib_send_event() already did the same. This is bad. When it's not in use we are not allowed to touch it. * Moved log message in front of the resulting actions and fixed it to match the case. Patches: patch5_bccleanup.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012) | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up etc., really bad stuff. * Fix return codes of cb_events() for EVENT_SETUP to use caller's cleanup mechanisms. * Move cl_queue_chan() call after bearer check. Patches: patch6_leaks.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374521 | rmudgett | 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines chan_misdn: We must initialize cause on sending a DISCONNECT. We must initialize cause on sending a DISCONNECT, so it is later correctly indicated to ast_channel in case the answer (RELEASE/RELEASE_COMPLETE) does not include one. Patches: patch7_hangupcause.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374522 | rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused code for upqueue Patches: patch8_unused-upqueue.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374523 | rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines chan_misdn: Improve debugging (port number, messages fixed, dups removed) Patches: patch9_debug.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2882 ................ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012) | 8 lines chan_misdn: Better debug: we can print_bc_info even if there's no ast leg. Patches: patch10_debug-bc-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2882 ................ r374534 | rmudgett | 2012-10-05 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn: setup_bc() is called too early for an incoming SETUP on TE. This prevents the B channel from being setup for HDLC mode when requested by the bearer capability and config option hdlc=yes. It violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not connect to the channel until a CONNECT ACKNOWLEDGE message has been received." * Call setup_bc() on receipt of CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by Guenther Kelleter Modified. JIRA ABE-2881 ................ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012) | 2 lines chan_misdn: Remove some more deadcode. ................ ........ Merged revisions 374536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374568 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	e40066e919	Merged revisions 374476,374481 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374476 | alecdavis | 2012-10-04 15:05:14 -0500 (Thu, 04 Oct 2012) | 13 lines dsp.c fix incorrect DTMF Digit_Duration. it's always short by 'hits_to_begin*DTMF_GSIZE', or 25.5ms if hitstobegin=2 (issue ASTERISK-16003) Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2145/ ........ Merged revisions 374475 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r374481 | alecdavis | 2012-10-04 15:17:16 -0500 (Thu, 04 Oct 2012) | 17 lines dsp.c User Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END Instead of a recompile, allow values to be adjusted in dsp.conf For binary distributions allows easy adjustment for wobbly GSM calls, and other reasons. Defaults to DTMF_HITS_TO_BEGIN=2 and DTMF_MISSES_TO_END=3 (closes issue ASTERISK-17493) Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2144/ ........ Merged revisions 374479 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	afa876c303	Merged revisions 374457 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374457 | file | 2012-10-04 12:44:38 -0500 (Thu, 04 Oct 2012) | 17 lines Fix a regression from direct media ACLs where the directrtpsetup option no longer works. A check was added for direct media ACLs that immediately forbid remote bridging if there was no bridged channel. This caused directrtpsetup to no longer function as it needs this information before bridging actually occurs. Logic has now been adjusted so if there is no bridged channel a remote bridge will still be attempted. (closes issue ASTERISK-20511) Reported by: kristoff Review: https://reviewboard.asterisk.org/r/2146/ ........ Merged revisions 374456 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374474 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	fa1312ae4c	Merged revisions 374427 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374427 | dlee | 2012-10-04 10:37:11 -0500 (Thu, 04 Oct 2012) | 25 lines Fix DBDelTree error codes for AMI, CLI and AGI The AMI DBDelTree command will return Success/Key tree deleted successfully even if the given key does not exist. The CLI command 'database deltree' had a similar problem, but was saved because it actually responded with '0 database entries removed'. AGI had a slightly different error, where it would return success if the database was unavailable. This came from confusion about the ast_db_deltree retval, which is -1 in the event of a database error, or number of entries deleted (including 0 for deleting nothing). * Changed some poorly named res variables to num_deleted * Specified specific errors when calling ast_db_deltree (database unavailable vs. entry not found vs. success) * Fixed similar bug in AGI database deltree, where 'Database unavailable' results in successful result (closes issue AST-967) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2138/ ........ Merged revisions 374426 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374455 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	e3f9a2b521	Merged revisions 374385 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374385 | alecdavis | 2012-10-03 23:41:19 -0500 (Wed, 03 Oct 2012) | 36 lines dsp.c User configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values Asterisk's DTMF Specifications are based on AT&T specs, which may not be compatible in other countries. Various countries have different specifications for the maximum power level differences between the DTMF low group and high group of frequencies. Power level difference between frequencies for different Administrations/RPOAs NTT = Max. 5 dB AT&T = 4dB(reverse) to 8dB(normal) Danish = Max. 6 dB Australian = Max. 10 dB Brazilian = Max. 9 dB ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1 (2006-03) Now allow 4 variables to be individually configured in dsp.conf, with reasonable min/max of 2dB to 20dB. Default is AT&T specifications Add's the following variables to dsp.conf ;dtmf_normal_twist=6.31 ;dtmf_reverse_twist=2.51 ;relax_dtmf_normal_twist=6.31 ;relax_dtmf_reverse_twist=3.98 (closes issue ASTERISK-20442) Reported by: tbsky Tested by: tbsky,alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2141/ ........ Merged revisions 374384 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	595c52a92c	Merged revisions 374370 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374370 | alecdavis | 2012-10-03 23:18:44 -0500 (Wed, 03 Oct 2012) | 15 lines _dsp_init: bring inline with trunk preparation for clean merge of DTMF TWIST patch No functional changes, just style. alecdavis (license 585) Reported by: Alec Davis Tested by: alecdavis related https://reviewboard.asterisk.org/r/2141 ........ Merged revisions 374365 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	32da9c5e91	Merged revisions 374336 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374336 | mjordan | 2012-10-03 21:11:05 -0500 (Wed, 03 Oct 2012) | 31 lines Check for presence of buddy in info/dinfo handlers The res_jabber resource module uses the ASTOBJ library for managing its ref counted objects. After calling ASTOBJ_CONTAINER_FIND to locate a buddy object, the pointer to the object has to be checked to see if the buddy existed. Prior to this patch, the buddy object was not checked for NULL; with this patch in both aji_client_info_handler and aji_dinfo_handler the pointer is checked before used and, if no buddy object was found, the handlers return an error code. This patch does not take the approach that our JID can be used to log in from another resource. If that approach is desired, an improvement could be made to this patch to create the buddy on the fly. This patch seeks only to prevent Asterisk from crashing. Note that multiple people have proposed patches for this issue; the patch being committed here is based on those. (closes issue ASTERISK-19532) Reported by: Karsten Wemheuer Tested by: Byron Clark patches: fix-jabber uploaded by Karsten Wemheuer (license #5930) xmpp_no_crash_with_ejabberd.patch uploaded by Byron Clark (license #6157) (closes issue ASTERISK-19557) Reported by: ulugutz ........ Merged revisions 374335 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	9faa31358a	Merged revisions 374317 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374317 | mjordan | 2012-10-03 13:59:06 -0500 (Wed, 03 Oct 2012) | 13 lines Blocked revisions 374316 ........ Destroy the generic_monitors container after the core_instances in ccss For each item in core_instances disposed of in the shutdown of ccss, any generic monitor instances referenced by the objects will be removed from generic_monitors during their destruction. Hilarity ensues if generic_monitors no longer exists. Thanks to the Asterisk Test Suite's generic_ccss test for complaining loudly when it ran into this. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374334 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	3be5589ab9	Merged revisions 374300 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r374300 | mjordan | 2012-10-03 12:25:36 -0500 (Wed, 03 Oct 2012) | 10 lines Destroy the generic_monitors container after the core_instances in ccss For each item in core_instances disposed of in the shutdown of ccss, any generic monitor instances referenced by the objects will be removed from generic_monitors during their destruction. Hilarity ensues if generic_monitors no longer exists. Thanks to the Asterisk Test Suite's generic_ccss test for complaining loudly when it ran into this. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374315 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	e469d6e0ce	Merged revisions 374231 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374231 | mjordan | 2012-10-02 16:12:30 -0500 (Tue, 02 Oct 2012) | 9 lines Ensure Shutdown AMI event is still fired during Asterisk shutdown Richard pointed out that having the manager dispose of itself gracefully during shutdown meant that the Shutdown event will no longer get fired. This patch moves the AMI event just prior to running the atexit callbacks. ........ Merged revisions 374230 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	8cb345c99e	Merged revisions 374210 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r374210 | mjordan | 2012-10-02 12:10:04 -0500 (Tue, 02 Oct 2012) | 10 lines Fix findings from check-in on r374177 Richard pointed out two problems with the check-in from r374177: * The ast_msg_shutdown function declaration doesn't match the prototype in main/message.c. * The ref/alloc function usage in astobj2 (in 11+) can use the ao2_t_* variants of the functions to allow the REF_DEBUG flag to enable/disable their debug counterparts. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	3409955b52	Re-enable automerge. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	8c55bd5dea	Fix a variety of ref counting issues ... This patch resolves a number of ref leaks that occur primarily on Asterisk shutdown. It adds a variety of shutdown routines to core portions of Asterisk such that they can reclaim resources allocate duringd initialization. Review: https://reviewboard.asterisk.org/r/2137 ........ Merged revisions 374177 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374178 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374208 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	b955a0c9ed	automerge cancel ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374195 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	322e224ec4	Merged revisions 374132,374135 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374132 | seanbright | 2012-10-01 12:27:22 -0500 (Mon, 01 Oct 2012) | 2 lines Use ast_copy_string instead of strncpy to guarantee a NUL terminated string. ................ r374135 | seanbright | 2012-10-01 12:52:38 -0500 (Mon, 01 Oct 2012) | 23 lines app_queue: Support persisting and loading of long member lists. Greenlight in #asterisk brought up that he was receiving an error message "Could not create persistent member string, out of space" when running app_queue in Asterisk 10. dump_queue_members() made an assumption that 8K would be enough to store the generated string, but with queues that have large member lists this is not always the case. This patch removes the limitation and uses ast_str instead of a fixed sized buffer. The complicating factor comes from the fact that ast_db_get requires a buffer and buffer size argument, which doesn't let us pull back more than what we pass in, so I introduced a new ast_db_get_allocated() which returns an ast_strdup()'d copy of the value from astdb. As an aside, I did some testing on the maximum size of data that we can store in the BDB library we distribute and was able to store a 10MB string and retrieve it with no problems, so I feel this is a safe patch. Review: https://reviewboard.asterisk.org/r/2136/ ........ Merged revisions 374108 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f796714b70	Merged revisions 374045 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r374045 | jrose | 2012-09-28 14:21:10 -0500 (Fri, 28 Sep 2012) | 12 lines res_jabber: Remove CLI command 'jabber test' The opinion of development was that it is both improper to have Matt's personal email address used in the source and that the command wouldn't be useful without it. (closes issue AST-467) Reported by: Malcolm Davenport ........ Merged revisions 374032 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374058 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	10036bbbe4	Merged revisions 373990 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373990 | file | 2012-09-28 07:15:48 -0500 (Fri, 28 Sep 2012) | 8 lines Update documentation to make it explicit that "stream file" will not restart musiconhold. (issue ASTERISK-17367) Reported by: oej ........ Merged revisions 373989 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@374018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	0915adaf03	Merged revisions 373946 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373946 | rmudgett | 2012-09-27 17:12:47 -0500 (Thu, 27 Sep 2012) | 14 lines Fix SendDTMF crash and channel reference leak using channel name parameter. The SendDTMF channel name parameter has two issues. 1) Crashes if the channel name does not exist. 2) Leaks a channel reference if the channel is the current channel. Problem introduced by ASTERISK-15956. * Updated SendDTMF documentation. * Renamed app to senddtmf_name and tweaked the type. ........ Merged revisions 373945 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	4ac9c29b8f	Merged revisions 373910 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373910 | file | 2012-09-27 11:50:46 -0500 (Thu, 27 Sep 2012) | 24 lines loader: Ensure dependent modules are properly initialized. If an Asterisk module specifies a dependency in ast_module_info.nonoptreq, it is possible for Asterisk to skip calling the modules's .load function. Asterisk was loading and linking the module via load_dynamic_module() but was not adding the module to the resource_heap. Therefore the module was not initialized based on it's priority along with the other modules in the heap. Now use load_resource() instead of load_dynamic_module() for non-optional requirement. This will add the module to the resource_heap so the module can be properly initialized in the correct order. This is required if there are any module global data structures initialized in the .load() callback for the module on platforms which do not support weak references. (issue ASTERISK-20439) Reported by: sruffell Patches: 0001-loader-Ensure-dependent-modules-are-properly-initial.patch uploaded by sruffell (license 5417) ........ Merged revisions 373909 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373941 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	00271653ec	Merged revisions 373879 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373879 | file | 2012-09-27 06:32:13 -0500 (Thu, 27 Sep 2012) | 14 lines Fix an issue where Local channels dialed by app_queue are considered in use immediately. The chan_local channel driver returns a device state of in use even if a created Local channel has not yet been dialed. This fix changes the logic to return a state of not in use until the channel itself has been dialed. (closes issue ASTERISK-20390) Reported by: tim_ringenbach Review: https://reviewboard.asterisk.org/r/2116/ ........ Merged revisions 373878 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373907 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	bfffa008f4	Merged revisions 373849 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373849 | mmichelson | 2012-09-26 16:11:35 -0500 (Wed, 26 Sep 2012) | 8 lines Move handling of 408 response so there is no misleading warning message. (closes issue ASTERISK-20060) Reported by: Walter Doekes ........ Merged revisions 373848 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373877 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f6ae5b7142	Merged revisions 373816 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373816 | rmudgett | 2012-09-26 13:15:50 -0500 (Wed, 26 Sep 2012) | 18 lines Fixed meetme tab completion and command documentation. * Removed unnecessary case sensitivity in meetme list, lock, unlock, mute, unmute, and kick commands. * Separated meetme lock/unlock, mute/unmute, and kick commands into their own registered commands to simplify tab completion and parameter checking. meetme_lock_cmd(), meetme_mute_cmd(), and meetme_kick_cmd() * Simplified meetme_show_cmd() (closes issue AST-1006) Reported by: John Bigelow Tested by: rmudgett ........ Merged revisions 373815 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	be949cda01	Merged revisions 373769,373774 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373769 | mmichelson | 2012-09-25 17:54:13 -0500 (Tue, 25 Sep 2012) | 11 lines Remove dead code and documentation for nonexistent feature. multiplelogin was removed from chan_agent back in 1.6.0 when AgentCallbackLogin() was removed. (closes issue AST-948) reported by Steve Pitts ........ Merged revisions 373768 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r373774 | mmichelson | 2012-09-25 18:08:46 -0500 (Tue, 25 Sep 2012) | 10 lines Fix saying of date in Dutch. The Dutch say the date before the month. (closes issue ASTERISK-20353) Reported by: Teun Ouwehand ........ Merged revisions 373773 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	6128a98e50	Merged revisions 373737 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373737 | mmichelson | 2012-09-25 16:12:40 -0500 (Tue, 25 Sep 2012) | 11 lines Fix error where improper IMAP greetings would be deleted. (closes issue ASTERISK-20435) Reported by: fhackenberger Patches: asterisk-20435-imap-del-greeting.diff uploaded by Michael L. Young (License #5026) (with suggested modification made by me) ........ Merged revisions 373735 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373765 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	e4231a9358	Merged revisions 373703,373706 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373703 | kmoore | 2012-09-25 14:34:01 -0500 (Tue, 25 Sep 2012) | 11 lines Fix an issue where media would not flow for situations where the legacy STUN code is in use. The STUN packets should *not* be blocked by strict RTP. (closes issue ASTERISK-20415) Reported-by: Michele Cicciotti Patch-by: Josh Colp (trunk r369817) ........ Merged revisions 373702 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r373706 | file | 2012-09-25 15:12:02 -0500 (Tue, 25 Sep 2012) | 22 lines Fix T.38 support when used with chan_local in between. Users of the T.38 API can indicate AST_T38_REQUEST_PARMS on a channel to request that the channel indicate a T.38 negotiation with the parameters present on the channel. The return value of this indication is expected to be AST_T38_REQUEST_PARMS upon success but with chan_local involved this could never occur. This fix changes chan_local to always return AST_T38_REQUEST_PARMS for this situation. If the underlying channel technology on the other side does not support T.38 this would have been determined ahead of time using ast_channel_get_t38_state and an indication would not occur. (closes issue ASTERISK-20229) Reported by: wdoekes Patches: ASTERISK-20229.patch uploaded by wdoekes (license 5674) Review: https://reviewboard.asterisk.org/r/2070/ ........ Merged revisions 373705 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	5dcc32ba72	Merged revisions 373665,373675 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373665 | twilson | 2012-09-25 12:35:30 -0500 (Tue, 25 Sep 2012) | 21 lines Properly handle UAC/UAS roles for SIP session timers The SIP session timer mechanism contains a mandatory 'refresher' parameter (included in the Session-Expires header) which is used in the session timer offer/answer signaling within a SIP Invite dialog. It looks like asterisk is interpreting the uac resp. uas role only as the initial role of client and server (caller is uac, callee is uas). The standard rfc 4028 however assigns the client role to the ((RE)-Invite) requester, the server role to the ((RE)-Invite) responder. This patch has Asterisk track the actual refresher as "us" or "them" as opposed to relying on just the configured "uas" or "uac" properties. (closes issue AST-922) Reported by: Thomas Airmont Review: https://reviewboard.asterisk.org/r/2118/ ........ Merged revisions 373652 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r373675 | kmoore | 2012-09-25 13:20:04 -0500 (Tue, 25 Sep 2012) | 13 lines "show" completion option for "queue" shouldn't appear twice When tab-completing CLI commands starting with "queue", "show" appeared twice in the list due to the way that Asterisk's tab completion functions and the order in which the commands were registered. The registration order has been altered to resolve this issue. (closes issue AST-940) Reported-by: Steve Pitts ........ Merged revisions 373666 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	f2608668e3	Merged revisions 373631,373633,373645 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373631 | jrose | 2012-09-25 11:24:34 -0500 (Tue, 25 Sep 2012) | 10 lines chan_sip: Set Quality of Service for video rtp instance (closes issue ASTERISK-20201) Reported by: ddkprog Patches: chan_sip.c.diff uploaded by ddkprog (license 6008) ........ Merged revisions 373617 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r373633 | rmudgett | 2012-09-25 11:33:31 -0500 (Tue, 25 Sep 2012) | 5 lines Make rebuild GSM, ilbc, or lpc10 codecs if the respective sources change. ........ Merged revisions 373618 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r373645 | rmudgett | 2012-09-25 12:19:52 -0500 (Tue, 25 Sep 2012) | 14 lines Fix valgrind found memcpy issues in codec_ilbc. Valgrind found codec_ilbc using memcpy instead of memmove for overlapping memory blocks. (issue ASTERISK-19890) (closes issue ASTERISK-20231) Reported by: Walter Doekes Patches: ASTERISK-20231.patch (license #5674) patch uploaded by Walter Doekes ........ Merged revisions 373640 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	7b9baa47f2	Merged revisions 373579 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373579 | kmoore | 2012-09-25 08:28:20 -0500 (Tue, 25 Sep 2012) | 11 lines Fix documentation for default username in res_odbc This was previously stated to be "root", but is actually the name of the context if unspecified. (closes issue ASTERISK-20258) Reported by: Stefan x ........ Merged revisions 373578 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	b822a6fcf7	Merged revisions 373551 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373551 | file | 2012-09-25 07:00:23 -0500 (Tue, 25 Sep 2012) | 15 lines Fix an issue where a caller to ast_write on a MulticastRTP channel would determine it failed when in reality it did not. When sending RTP packets via multicast the amount of data sent is stored in a variable and returned from the write function. This is incorrect as any non-zero value returned is considered a failure while a return value of 0 is success. For callers (such as ast_streamfile) that checked the return value they would have considered it a failure when in reality nothing went wrong and it was actually a success. The write function for the multicast RTP engine now returns -1 on failure and 0 on success, as it should. (closes issue ASTERISK-17254) Reported by: wybecom ........ Merged revisions 373550 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	ef7b6d6adc	Merged revisions 373533 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373533 | file | 2012-09-24 19:11:28 -0500 (Mon, 24 Sep 2012) | 5 lines Add missing checks that I neglected. The SIP technology and SIP info technology should be considered equal. ........ Merged revisions 373532 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373549 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	b6142804e4	Merged revisions 373501,373505 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373501 | rmudgett | 2012-09-24 17:11:01 -0500 (Mon, 24 Sep 2012) | 18 lines Be consistent, send From: "Anonymous" <sip:anonymous@anonymous.invalid> When setting CALLERID(pres)=unavailable in the dialplan, the From header in the SIP message contains "Anonymous" <sip:Anonymous@anonymous.invalid>. For consistency, Asterisk should use a lowercase a in the userpart of the URI. * Make the From header use a lowercase A in the userpart of the anonymous URI. (closes issue ASTERISK-19838) Reported by: Antti Yrjola Patches: chan_sip_patch_ASTERISK-19838.patch (license #6383) patch uploaded by Antti Yrjola ........ Merged revisions 373500 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r373505 | mjordan | 2012-09-24 17:17:02 -0500 (Mon, 24 Sep 2012) | 19 lines Revert change to res_rtp_asterisk committed in r373236 (1.8) The change committed in r373236 attempted to account for endpoints that increased their RTP timestamp in DTMF end of event re-transmissions. This change attempted to make Asterisk continue to work with endpoints that failed to follow the RFC while maintaining the fix that allowed for out of order DTMF to be handled. Unfortunately, there is no free lunch, and this patch broke any system that sent DTMF immediately after an RTP session was established or when an SSRC is updated. As such, that patch is being reverted for the previous behavior. Endpoints that erroneously increase the RTP timestamp in DTMF end of event packets will not work properly with Asterisk. (issue ASTERISK-20424) ........ Merged revisions 373504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	3123eb71c7	Merged revisions 373466,373468 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373466 | rmudgett | 2012-09-24 15:44:27 -0500 (Mon, 24 Sep 2012) | 33 lines Fix potential reentrancy problems in chan_sip. Asterisk v1.8 and later was not as vulnerable to this issue. * Made find_call() lock each private as it processes the found dialogs. (Primary cause of ABE-2876) * Made the other functions that traverse the dialogs container lock each private as it examines them. * Fix race condition in sip_call() if the thread that sent the INVITE is held up long enough for a response to be processed. The p->initid for the INVITE retransmission could be added after it was canceled by the response processing. * Made __sip_destroy() clean up resource pointers after freeing. This is primarily defensive in case someone has a stale private pointer. * Removed redundant memset() in reqprep(). The call to init_req() already does the memset() and is the first reference to req in reqprep(). * Removed useless set of req.method in transmit_invite(). The calls to initreqprep() and reqprep() have to do this because they memset() the req. JIRA ABE-2876 .......... Merged -r373423 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 373424 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r373468 | jrose | 2012-09-24 16:05:44 -0500 (Mon, 24 Sep 2012) | 13 lines func_audiohookinherit: Document some missed sources. This patch also mentions that AUDIOHOOK_INHERIT can be used to transfer MixMonitor audiohooks. There is also wiki that addresses audiohooks and the use of AUDIOHOOK_INHERIT at the following link: https://wiki.asterisk.org/wiki/display/AST/Audiohooks (closes issue ASTERISK-18220) Reported by: Ishfaq Malik ........ Merged revisions 373467 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373498 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	a48d4a2c84	reenable automerge ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373491 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	efdc1001e8	Fix a deadlock caused by a race condition between removing a hint and reloading the dialplan and subscribing to the removed hint. ... If conditions were right it was possible for both the PBX core and chan_sip to deadlock by both having a lock that the other wants. In the case of the PBX core it had the contexts lock and wanted a SIP dialog lock, while in the case of chan_sip it had the SIP dialog lock and wanted the contexts lock. This fix unlocks the SIP dialog before getting the extension state so that the other thread will not block on trying to lock it. Once the extension state is retrieved the SIP dialog is locked again and life carries on. As the SIP dialog is reference counted it is not possible for it to go away after unlocking. (closes issue ASTERISK-20437) Reported by: jhutchins ........ Merged revisions 373438 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373440 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	3ff80fa671	automerge cancel ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373455 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	d1500e8924	Merged revisions 373343 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373343 | jrose | 2012-09-21 14:08:58 -0500 (Fri, 21 Sep 2012) | 10 lines iax2-provision: Fix improper return on failed cache retrieval (closes issue ASTERISK-20337) reported by: John Covert Patches: iax2-provision.c.patch uploaded by John Covert (license 5512) ........ Merged revisions 373342 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	c11c20e464	Merged revisions 373300 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373300 | jrose | 2012-09-21 10:07:38 -0500 (Fri, 21 Sep 2012) | 12 lines app_queue: Make queue reload members and variants of that work Prior to this patch, 'queue reload members' cli command did not work at all. This also affects the manager function 'QueueReload' when supplied with the 'members: yes' field. (closes issue AST-956) Reported by: John Bigelow ........ Merged revisions 373298 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	22c0ba7287	Merged revisions 373237,373245 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ................ r373237 | mjordan | 2012-09-20 13:42:51 -0500 (Thu, 20 Sep 2012) | 18 lines When processing RFC 2833 DTMF, accomodate increasing timestamps in End events While endpoints should not be changing the source timestamp between DTMF event packets, the fact is there exists those endpoints that do exactly that. To work around this, we absorb timestamps within the expected re-transmit period. Note that this period only affects End of Event packets, so it should not prevent the detection of new DTMF digits that happen to arrive right on top of each other. (closes issue ASTERISK-20424) Reported by: Vladimir Mikhelson Tested by: mjordan, Vladimir Mikhelson Review: https://reviewboard.asterisk.org/r/2124 ........ Merged revisions 373236 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ r373245 | file | 2012-09-20 14:14:31 -0500 (Thu, 20 Sep 2012) | 15 lines Fix incorrect MeetME conference bridge reference count decrementing and sometimes premature destruction. When using the 'e' or 'E' option to MeetMe the configured conference bridges are loaded and examined to see if any are empty. If no conference bridges are empty the caller is prompted to enter the number of one. This operation left around a pointer to the last created conference bridge still containing participants. When the caller that was not able to find any empty conference bridge hung up this pointer was disposed of and the reference count of the conference bridge decremented. If there was only a single participant in the conference bridge it was ultimately destroyed prematurely. (closes issue AST-994) Reported by: John Bigelow ........ Merged revisions 373242 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373274 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Automerge script	d26918bd63	Merged revisions 373196 via svnmerge from ... file:///srv/subversion/repos/asterisk/branches/10 ........ r373196 | mjordan | 2012-09-19 21:35:13 -0500 (Wed, 19 Sep 2012) | 12 lines Ensure that all ConfBridge sounds can be set using CONFBRIDGE function The CONFBRIDGE function can be used to set the sounds in a ConfBridge bridge profile. Unfortunately, three sounds were missed in the portion of the code that applies the settings passed in from the function: sound_only_one, join, and leave. This patch makes sure that the sounds passed from the function are applied to the bridge profile. (closes issue ASTERISK-20404) Reported by: univ Tested by: mjordan ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago