asterisk

Commit Graph

Author	SHA1	Message	Date
Mark Michelson	d24e3e05bc	Merged revisions 180007 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180007 \| mmichelson \| 2009-03-03 16:49:07 -0600 (Tue, 03 Mar 2009) \| 22 lines Merged revisions 180006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180006 \| mmichelson \| 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) \| 17 lines Clarify some documentation of queues.conf.sample It had always been possible to explicitly specify a "blank" value for a sound file in queues.conf and have no sound played back. The problem with this is that it would result in some ugly CLI warnings from file.c. This commit introduces a check when playing a file in app_queue to see if the name of the file is zero-length and return early if that is the case. Also, the ability to specify the blank sound files in queues.conf is now mentioned more clearly in queues.conf.sample (closes issue #14227) Reported by: caspy ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	3cb51f0127	Fix a memory leak when updating a realtime member field. This was discovered while looking at issue #14353 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179971 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	49d2383e12	Merged revisions 179533 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179533 \| russell \| 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009) \| 48 lines Merged revisions 179532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179532 \| russell \| 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) \| 40 lines Move ast_waitfor() down to avoid the results of the API call becoming stale. This call to ast_waitfor() was being done way too soon in this section of code. Specifically, there was code in between the call to waitfor and the code that uses the result that puts the channel in autoservice. By putting the channel in autoservice, the previous results of ast_waitfor() become meaningless, as the autoservice thread will do it's own ast_waitfor() and ast_read() on the channel. So, when we came back out of autoservice and eventually hit the block of code that calls ast_read() on the channel, there may not actually be any input on the channel available. Even though the previous call to ast_waitfor() in app_meetme said there was input, the autoservice thread has since serviced the channel for some period of time. This bug manifested itself while dvossel was doing some testing of MeetMe in Asterisk trunk. He was using the timerfd timing module. When the code hit ast_read() erroneously, it determined that it must have been called because of input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was the cause of the last legitimate call to ast_read() done by autoservice. In this test, an IAX2 channel was calling into the MeetMe conference. It was _much_ more likely to be seen with an IAX2 channel because of the way audio is handled. Every audio frame that comes in results in a call to ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify the channel thread that a frame is waiting to be handled. So, the chances of ast_waitfor() indicating that a channel needs servicing due to a timer event on an IAX2 event is very high. Finally, it is interesting to note that if a different timing interface was being used, this bug would probably not be noticed. When ast_read() is called and erroneously thinks that there is a timer event to handle, it calls the ast_timer_ack() function. The pthread and dahdi timing modules handle the ack() function being called when there is no event by simply ignoring it. In the case of the timerfd module, it results in a read() on the timer fd that will block forever, as there is no data to read. This caused Asterisk to lock up very quickly. Thanks to dvossel and mmichelson for the fun debugging session. :-) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	33d3f5ab4f	KeepAlive application no longer exists, so fix gosub implementation to not use it. (closes issue #14571) Reported by: zktech Patches: 20090302__bug14571.diff.txt uploaded by tilghman (license 14) Tested by: tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	eb9e2f2c4e	Add error checking when updating the "paused" field of a realtime queue member. This code already existed in trunk and 1.6.1, but was not in 1.6.0 prior to this commit. (closes issue #14338) Reported by: fiddur Patches: 14338.patch uploaded by mmichelson (license 60) Tested by: fiddur git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179222 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	de2bbd8f13	Change include order to make compile on Centos 5 with DAHDI If BIT_TYPES_DEFINED gets defined before linux/types.h is included, the __s32 type doesn't get defined git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178306 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Dwayne M. Hubbard	e8073da399	Merged revisions 177699 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177699 \| dhubbard \| 2009-02-20 14:29:00 -0600 (Fri, 20 Feb 2009) \| 9 lines Make app_fax compatible with spandsp-0.0.6pre4 Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a pages_transferred integer to indicate the number of pages transferred (so far) during the fax session. The spandsp-0.0.6pre4 release removed the pages_transferred integer and replaced it with two different integers - pages_tx and pages_rx. This revision uses the new integers for spandsp-0.0.6pre4 while maintaining backwards compatibility for previous spandsp releases. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177763 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Dwayne M. Hubbard	14bb8408db	Merged revisions 148868 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r148868 \| tilghman \| 2008-10-14 10:15:45 -0500 (Tue, 14 Oct 2008) \| 6 lines API differences in spandsp 0.0.6pre1 and higher (closes issue #13688) Reported by: irroot Patches: app_fax-span6.patch uploaded by irroot (license 52) with minor modifications by me ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	36f41cee56	Merged revisions 177664 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177664 \| tilghman \| 2009-02-20 11:29:51 -0600 (Fri, 20 Feb 2009) \| 8 lines Allow semicolons to be escaped, when passing arguments to the System command. (closes issue #14231) Reported by: jcovert Patches: 20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14) corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551) Tested by: jcovert ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	b2cc0480ac	Merged revisions 177661 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177661 \| tilghman \| 2009-02-20 11:22:19 -0600 (Fri, 20 Feb 2009) \| 2 lines Oops, merge broke trunk ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177662 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8b4c344ca7	Merged revisions 177537 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177537 \| tilghman \| 2009-02-19 16:33:00 -0600 (Thu, 19 Feb 2009) \| 14 lines Merged revisions 177536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177536 \| tilghman \| 2009-02-19 16:26:01 -0600 (Thu, 19 Feb 2009) \| 7 lines Fix up potential crashes, by reducing the sharing between interactive and non-interactive threads. (closes issue #14253) Reported by: Skavin Patches: 20090219__bug14253.diff.txt uploaded by Corydon76 (license 14) Tested by: Skavin ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	d82a4d919c	Merged revisions 177384 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177384 \| file \| 2009-02-19 12:38:41 -0400 (Thu, 19 Feb 2009) \| 10 lines Merged revisions 177383 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177383 \| file \| 2009-02-19 12:37:25 -0400 (Thu, 19 Feb 2009) \| 3 lines If we are able to create a speech structure unset the ERROR variable in case it was previously set. (issue #LUMENVOX-13) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	ffa0eaebef	Merged revisions 177101 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177101 \| russell \| 2009-02-18 13:12:49 -0600 (Wed, 18 Feb 2009) \| 8 lines Re-add 'o' option to MeetMe, reverting rev 62297. Enabling this option by default proved to be a bad idea, as the talker detection is not very reliable. So, make it optional again, and off by default. (issue #13801) Reported by: justdave ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177135 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	d504f89a9d	Merged revisions 176253 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176253 \| mmichelson \| 2009-02-16 15:40:40 -0600 (Mon, 16 Feb 2009) \| 24 lines Merged revisions 176249,176252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176249 \| mmichelson \| 2009-02-16 15:34:27 -0600 (Mon, 16 Feb 2009) \| 14 lines Open the DAHDI pseudo device and set it to be nonblocking atomically Apparently on FreeBSD, attempting to set the O_NONBLOCKING flag separately from opening the file was causing an "inappropriate ioctl for device" error. While I cannot fathom why this would be happening, I certainly am not opposed to making the code a bit more compact/efficient if it also fixes a bug. (closes issue #14482) Reported by: ys Patches: meetme.patch uploaded by ys (license 281) Tested by: ys ........ r176252 \| mmichelson \| 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) \| 3 lines Remove unused variable and make dev-mode compilation happy ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	496e168b87	Merged revisions 175549 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175549 \| file \| 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) \| 4 lines Add an option to keep the recorded file upon hangup. (closes issue #14341) Reported by: fnordian ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	b8356f9a94	Merged revisions 174948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174948 \| mmichelson \| 2009-02-11 17:03:08 -0600 (Wed, 11 Feb 2009) \| 35 lines Fix odd "thank you" sound playing behavior in app_queue.c If someone has configured the queue to play an position or holdtime announcement, then it is odd and potentially unexpected to hear a "Thank you for your patience" sound when no position or holdtime was actually announced. This fixes the announcement so that the "thanks" sound is only played in the case that a position or holdtime was actually announced. There is a way that the "thank you" sound can be played without a position or holdtime, and that is to set announce-frequency to a value but keep announce-position and announce-holdtime both turned off. (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch uploaded by putnopvut (license 60) Tested by: caspy ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	a45ec0c30a	Merged revisions 174945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174945 \| mmichelson \| 2009-02-11 16:41:01 -0600 (Wed, 11 Feb 2009) \| 29 lines Fix 'd' option for app_dial and add new option to Answer application The 'd' option would not work for channel types which use RTP to transport DTMF digits. The only way to allow for this to work was to answer the channel if we saw that this option was enabled. I realized that this may cause issues with CDRs, specifically with giving false dispositions and answer times. I therefore modified ast_answer to take another parameter which would tell if the CDR should be marked answered. I also extended this to the Answer application so that the channel may be answered but not CDRified if desired. I also modified app_dictate and app_waitforsilence to only answer the channel if it is not already up, to help not allow for faulty CDR answer times. All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the changes except for the change to the Answer application will go in since we do not introduce new features into stable branches (closes issue #14164) Reported by: DennisD Patches: 14164.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/145 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	ae6b71dfb6	Merged revisions 174805 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174805 \| mmichelson \| 2009-02-10 17:17:03 -0600 (Tue, 10 Feb 2009) \| 11 lines Fix potential for stack overflows in app_chanspy.c When using the 'g' or 'e' options, the stack allocations that were used could cause a stack overflow if a spyer stayed on the line long enough without actually successfully spying on anyone. The problem has been corrected by using static buffers and copying the contents of the appropriate strings into them instead of using functions like alloca or ast_strdupa ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174820 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Steve Murphy	707028163d	For some strange reason, I didn't think 1.6.0 needed this fix. I was wrong. Here it is. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174439 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	e7a195d88b	Merged revisions 173773 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173773 \| mmichelson \| 2009-02-05 17:28:19 -0600 (Thu, 05 Feb 2009) \| 7 lines Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage (closes issue #13905) Reported by: jaroth Patches: foldermove_v2.patch uploaded by jaroth (license 50) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173774 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	4bd27c1d11	Merged revisions 173697 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173697 \| jpeeler \| 2009-02-05 15:00:26 -0600 (Thu, 05 Feb 2009) \| 18 lines Merged revisions 173696 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173696 \| jpeeler \| 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) \| 12 lines Add new configuration option to make shared IMAP mailboxes function as expected. The new option is "imapvmshareid" which is an ID to tag multiple mailboxes using the same IMAP storage location to function as one mailbox. This allows all messages to be retrieved for any user in the group. The patch alters the 'X-Asterisk-VM-Extension' header that is responsible for matching voicemails for a given user. (closes issue #13673) Reported by: howardwilkinson ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	a31c0961ab	Merged revisions 173693 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173693 \| mmichelson \| 2009-02-05 14:30:45 -0600 (Thu, 05 Feb 2009) \| 20 lines Merged revisions 173692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173692 \| mmichelson \| 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) \| 12 lines Fix situations where queue members could be autopaused unexpectedly Specifically, this patch prevents us from autopausing members when we receive a busy or congestion frame from them. (closes issue #14376) Reported by: fiddur Patches: 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173694 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	1efdb3072c	Merged revisions 173593 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173593 \| mmichelson \| 2009-02-05 12:48:55 -0600 (Thu, 05 Feb 2009) \| 11 lines Merged revisions 173592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173592 \| mmichelson \| 2009-02-05 12:47:24 -0600 (Thu, 05 Feb 2009) \| 3 lines Add some missing cleanup to app_mixmonitor ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173594 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	32f3ed9929	Merged revisions 173589 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173589 \| mmichelson \| 2009-02-05 12:34:06 -0600 (Thu, 05 Feb 2009) \| 33 lines Merged revisions 173559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173559 \| mmichelson \| 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) \| 25 lines Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up. app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed audio to a file. Since this thread runs independently of the channel, it is possible that the mixmonitor thread's channel pointer will point to freed memory when the channel either is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the cases slightly differently). The solution for this is to employ a datastore, which has the nice benefit of allowing us to hook into channel masquerades and hangups and update our pointer as necessary. If this looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more involved since it does a lot more operations on the channel that is being spied upon. app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em- ploy a condition-and-boolean combination to ensure that the channel thread finishes with our structure before the mixmonitor thread attempts to free it. No crashes! (closes issue #14374) Reported by: aragon Patches: 14374.patch uploaded by putnopvut (license 60) Tested by: aragon, putnopvut ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173590 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	f94677ac39	Merged revisions 173507 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173507 \| mmichelson \| 2009-02-04 16:16:19 -0600 (Wed, 04 Feb 2009) \| 7 lines Fix some areas where the incorrect interface was passed to ast_device_state I swear it feels like I already did this once... (closes issue #14359) Reported by: francesco_r ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	826418a799	Merged revisions 173397 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173397 \| mmichelson \| 2009-02-04 11:45:14 -0600 (Wed, 04 Feb 2009) \| 11 lines Merged revisions 173396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173396 \| mmichelson \| 2009-02-04 11:44:48 -0600 (Wed, 04 Feb 2009) \| 3 lines Revert my previous change because it was stupid ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173398 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	e1136de5bf	Merged revisions 173393 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173393 \| mmichelson \| 2009-02-04 11:41:02 -0600 (Wed, 04 Feb 2009) \| 11 lines Merged revisions 173392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173392 \| mmichelson \| 2009-02-04 11:40:29 -0600 (Wed, 04 Feb 2009) \| 3 lines Add a missing unlock. Extremely unlikely to ever matter, but it's needed. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@173394 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	66d4b791c8	Merged revisions 172741 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172741 \| tilghman \| 2009-01-31 20:44:23 -0600 (Sat, 31 Jan 2009) \| 4 lines Blank argument crashes Asterisk (closes issue #14377) Reported by: amorsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172742 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	af2b34cb56	Merged revisions 172580 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r172580 \| twilson \| 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009) \| 44 lines Merged revisions 172517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172517 \| twilson \| 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) \| 37 lines Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	407d3d8861	Merged revisions 172441 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r172441 \| tilghman \| 2009-01-29 17:15:40 -0600 (Thu, 29 Jan 2009) \| 16 lines Merged revisions 172438 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172438 \| tilghman \| 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) \| 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if Asterisk runs as a non-root user and the administrator does a 'restart now', Asterisk loses the ability to set QOS on packets. (closes issue #14004) Reported by: nemo Patches: 20090105__bug14004.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172503 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	491c4a9c68	Merged revisions 172063 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r172063 \| murf \| 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) \| 52 lines Merged revisions 172030 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172030 \| murf \| 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) \| 46 lines This patch fixes h-exten running misbehavior in manager-redirected situations. What it does: 1. A new Flag value is defined in include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the bridge hangup exten code not to run the h-exten there (nor publish the bridge cdr there). It will done at the pbx-loop level instead. 2. In the manager Redirect code, I set this flag on the channel if the channel has a non-null pbx pointer. I did the same for the second (chan2) channel, which gets run if name2 is set... and the first succeeds. 3. I restored the ending of the cdr for the pbx loop h-exten running code. Don't know why it was removed in the first place. 4. The first attempt at the fix for this bug was to place code directly in the async_goto routine, which was called from a large number of places, and could affect a large number of cases, so I tested that fix against a fair number of transfer scenarios, both with and without the patch. In the process, I saw that putting the fix in async_goto seemed not to affect any of the blind or attended scenarios, but still, I was was highly concerned that some other scenarios I had not tested might be negatively impacted, so I refined the patch to its current scope, and jmls tested both. In the process, tho, I saw that blind xfers in one situation, when the one-touch blind-xfer feature is used by the peer, we got strange h-exten behavior. So, I inserted code to swap CDRs and to set the HANGUP_DONT field, to get uniform behavior. 5. I added code to the bridge to obey the HANGUP_DONT flag, skipping both publishing the bridge CDR, and running the h-exten; they will be done at the pbx-loop (higher) level instead. 6. I removed all the debug logs from the patch before committing. 7. I moved the AUTOLOOP set/reset in the h-exten code in res_features so it's only done if the h-exten is going to be run. A very minor performance improvement, but technically correct. (closes issue #14241) Reported by: jmls Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17) Tested by: murf, jmls ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172065 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	344dfe6f84	Merged revisions 171618 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r171618 \| mmichelson \| 2009-01-27 13:30:54 -0600 (Tue, 27 Jan 2009) \| 24 lines Fix queue crashes that would occur after the calling channel was masqueraded. The data passed to the end_bridge_callback was assumed to be data which was still stack'd. The problem was that with some call features, attended transfers in particular, a new bridge thread is started once the feature completes, meaning that when the end_bridge_callback is called, the end_bridge_callback_data was invalid. To fix this problem, there are two measures taken 1. Instead of pointing to stacked data, we now used heap-allocated data for passing to the end_bridge_callback in app_queue 2. Since bridges can end multiple times on a single logical call, we wait until the final bridge is broken to actually set any queue variables. This is accomplished through reference-counting and the use of an end_bridge_callback_data_fixup function in app_queue.c (closes issue #14260) Reported by: ccesario Patches: 14260.patch uploaded by putnopvut (license 60) Tested by: ccesario ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@171619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	ecaaf9ca05	Merged revisions 170980 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170980 \| seanbright \| 2009-01-25 08:35:48 -0500 (Sun, 25 Jan 2009) \| 16 lines Merged revisions 170979 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170979 \| seanbright \| 2009-01-25 08:33:20 -0500 (Sun, 25 Jan 2009) \| 9 lines Resolve a logic error that was causing Page() to crash when more than one channel was specified. (closes issue #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt uploaded by seanbright (license 71) Tested by: kc0bvu ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170981 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	d00bc0014d	Merged revisions 170569 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170569 \| file \| 2009-01-23 15:09:18 -0400 (Fri, 23 Jan 2009) \| 11 lines Merged revisions 170568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170568 \| file \| 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) \| 4 lines When a call is forwarded stop any active indications. The new channel will provide an indication, if need be, itself. (closes issue #14310) Reported by: RadicAlish ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	cb0c483da0	Merged revisions 170148 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170148 \| file \| 2009-01-22 12:52:21 -0400 (Thu, 22 Jan 2009) \| 11 lines Merged revisions 170147 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170147 \| file \| 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) \| 4 lines If we are unable to request a DAHDI pseudo channel and we are using the user introduction without review option make sure it gets unset so other code does not blindly assume a DAHDI pseudo channel exists. (closes issue #14282) Reported by: cheesegrits ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170149 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	7766df456f	Merged revisions 170047 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r170047 \| file \| 2009-01-22 11:01:54 -0400 (Thu, 22 Jan 2009) \| 4 lines Clear the autoloop flag when parsing and setting the context/extension/priority to go back to. When the channel executes a PBX again we want it to start out at the point we explicitly say and at that point it will not yet be doing autoloop. (closes issue #14304) Reported by: jcovert ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	827aab8b29	Merged revisions 169611 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169611 \| mmichelson \| 2009-01-20 18:33:32 -0600 (Tue, 20 Jan 2009) \| 22 lines Fix device state parsing issues for channel names with multiple slashes The fix being applied is a bit different for trunk and the 1.6.X branches. For trunk, we only wish to strip off the characters beyond the second slash if the channel is a Local channel (i.e. we are removing the /n from the device name). Other channel technologies with multiple slashes (e.g. DAHDI) need the information after the second slash in order to get the proper device state information. In addition to this fix, the 1.6.X branches are receiving a much more important fix as well. The problem in 1.6.X is that the member's device name was being directly changed instead of having a copy changed. This meant that we would strip off the second slash and trailing characters and then leave the member's device name like that permanently thereafter. (closes issue #14014) Reported by: kebl0155 Patches: 14014_number2.patch uploaded by putnopvut (license 60) Tested by: kebl0155 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	0a108b6c1c	Merged revisions 169365 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r169365 \| tilghman \| 2009-01-19 14:05:52 -0600 (Mon, 19 Jan 2009) \| 11 lines Merged revisions 169364 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169364 \| tilghman \| 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009) \| 4 lines Truncate userevents at the end of a line, when the command exceeds the buffer. (closes issue #14278) Reported by: fnordian ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169366 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	b669fac978	Merged revisions 168832 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168832 \| tilghman \| 2009-01-16 12:49:09 -0600 (Fri, 16 Jan 2009) \| 13 lines Merged revisions 168828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168828 \| tilghman \| 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009) \| 6 lines Fix the conjugation of Russian and Ukrainian languages. (related to issue #12475) Reported by: chappell Patches: vm_multilang.patch uploaded by chappell (license 8) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	bd1a0134d6	Merged revisions 168705 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168705 \| seanbright \| 2009-01-15 10:33:18 -0500 (Thu, 15 Jan 2009) \| 11 lines Add a missing unlock and properly handle the 'maxusers' setting on MeetMe conferences. We were using the 'user number' field to compare against the maximum allowed users, which works assuming users with lower user numbers didn't leave the conference. (closes issue #14117) Reported by: sergedevorop Patches: 20090114__bug14117-2.diff.txt uploaded by seanbright (license 71) Tested by: sergedevorop ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	b40ebef036	Merged revisions 168629 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168629 \| mmichelson \| 2009-01-14 18:14:17 -0600 (Wed, 14 Jan 2009) \| 24 lines Merged revisions 168628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168628 \| mmichelson \| 2009-01-14 18:11:01 -0600 (Wed, 14 Jan 2009) \| 16 lines Fix some crashes from bad datastore handling in app_queue.c * The queue_transfer_fixup function was searching for and removing the datastore from the incorrect channel, so this was fixed. * Most datastore operations regarding the queue_transfer datastore were being done without the channel locked, so proper channel locking was added, too. (closes issue #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by putnopvut (license 60) Tested by: ZX81, festr ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168630 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	240cd8d0be	Merged revisions 168613 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168613 \| murf \| 2009-01-14 13:51:26 -0700 (Wed, 14 Jan 2009) \| 9 lines Merged revisions 168608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168608 \| murf \| 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) \| 1 line app_page was failing to compile in dev-mode on my gcc-4.2.4 system. This change gets rid of the warning. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	cee2699a5d	Merged revisions 168594 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168594 \| twilson \| 2009-01-13 20:00:40 -0600 (Tue, 13 Jan 2009) \| 27 lines Merged revisions 168593 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168593 \| twilson \| 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) \| 20 lines Don't overflow when paging more than 128 extensions The number of available slots for calls in app_page was hardcoded to 128. Proper bounds checking was not in place to enforce this limit, so if more than 128 extensions were passed to the Page() app, Asterisk would crash. This patch instead dynamically allocates memory for the ast_dial structures and removes the (non-functional) arbitrary limit. This issue would have special importance to anyone who is dynamically creating the argument passed to the Page application and allowing more than 128 extensions to be added by an outside user via some external interface. The patch posted by a_villacis was slightly modified for some coding guidelines and other cleanups. Thanks, a_villacis! (closes issue #14217) Reported by: a_villacis Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660) Tested by: otherwiseguy ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	1b1c2db6bd	Merged revisions 168562 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168562 \| russell \| 2009-01-13 13:22:13 -0600 (Tue, 13 Jan 2009) \| 10 lines Merged revisions 168561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 \| russell \| 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) \| 2 lines Revert unnecessary indications API change from rev 122314 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168564 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	eeac5f539e	Merged revisions 168497 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168497 \| oej \| 2009-01-12 17:31:27 +0100 (MÃ¥n, 12 Jan 2009) \| 2 lines Better to use the proper app name in the STATUS variable ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168499 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	7970bef2d2	Merged revisions 167935 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r167935 \| twilson \| 2009-01-08 18:13:12 -0600 (Thu, 08 Jan 2009) \| 2 lines Set peer context and exten values so MACRO_EXTEN and MACRO_CONTEXT will be set ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@167971 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
BJ Weschke	976f09e010	Merged revisions 167478 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r167478 \| bweschke \| 2009-01-07 13:20:31 -0500 (Wed, 07 Jan 2009) \| 7 lines Answer the channel if it has not already been answered and we've already found a valid profile for followme. (closes issue #14140) Reported by: dimas Patches: 14140.patch uploaded by dimas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@167498 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	788f3a8fa5	Merged revisions 166861 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166861 \| mmichelson \| 2008-12-29 12:04:52 -0600 (Mon, 29 Dec 2008) \| 14 lines Update app_queue to deal with the removal of AST_PBX_KEEPALIVE When placing a call to a queue which ran a gosub on the member's channel, Asterisk would crash every time, stemming from the fact that the member's channel was being hung up unexpectedly when the Gosub completed. The necessary change was pretty much copied and pasted from app_dial's similar changes made last week. I also took the opportunity to change a LOG_DEBUG message in app_dial to use ast_debug. I am guessing this was due to a direct merge from 1.4 that was not corrected to use trunk's preferred syntax. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166862 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	5d34e0df03	Merged revisions 166665 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk Due to non-symmetrical updating, I had some fairly interesting conflicts to straighten out in this release. The changes were such that I was compelled to run thru all the same tests as trunk, which turned up some problems, which I fixed. ................ r166665 \| murf \| 2008-12-23 11:13:49 -0700 (Tue, 23 Dec 2008) \| 153 lines Merged revisions 166093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In order to merge this 1.4 patch into trunk, I had to resolve some conflicts and wait for Russell to make some changes to res_agi. I re-ran all the tests; 39 calls in all, and made fairly careful notes and comparisons: I don't want this to blow up some aspect of asterisk; I completely removed the KEEPALIVE from the pbx.h decls. The first 3 scenarios involving feature park; feature xfer to 700; hookflash park to Park() app call all behave the same, don't appear to leave hung channels, and no crashes. ........ r166093 \| murf \| 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) \| 131 lines This merges the masqpark branch into 1.4 These changes eliminate the need for (and use of) the KEEPALIVE return code in res_features.c; There are other places that use this result code for similar purposes at a higher level, these appear to be left alone in 1.4, but attacked in trunk. The reason these changes are being made in 1.4, is that parking ends a channel's life, in some situations, and the code in the bridge (and some other places), was not checking the result code properly, and dereferencing the channel pointer, which could lead to memory corruption and crashes. Calling the masq_park function eliminates this danger in higher levels. A series of previous commits have replaced some parking calls with masq_park, but this patch puts them ALL to rest, (except one, purposely left alone because a masquerade is done anyway), and gets rid of the code that tests the KEEPALIVE result, and the NOHANGUP_PEER result codes. While bug 13820 inspired this work, this patch does not solve all the problems mentioned there. I have tested this patch (again) to make sure I have not introduced regressions. Crashes that occurred when a parked party hung up while the parking party was listening to the numbers of the parking stall being assigned, is eliminated. These are the cases where parking code may be activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via manager. The interesting testing cases for parking are: I. A calls B, A parks B a. B hangs up while A is getting the numbers announced. b. B hangs up after A gets the announcement, but before the parking time expires c. B waits, time expires, A is redialed, A answers, B and A are connected, after which, B hangs up. d. C picks up B while still in parking lot. II. A calls B, B parks A a. A hangs up while B is getting the numbers announced. b. A hangs up after B gets the announcement, but before the parking time expires c. A waits, time expires, B is redialed, B answers, A and B are connected, after which, A hangs up. d. C picks up A while still in parking lot. Testing this throroughly involves acting all the permutations of I and II, in situations 1,2,3, and 4. Since I added a few more changes (ALL references to KEEPALIVE in the bridge code eliimated (I missed one earlier), I retested most of the above cases, and no crashes. H-extension weirdness. Current h-extension execution is not completely correct for several of the cases. For the case where A calls B, and A parks B, the 'h' exten is run on A's channel as soon as the park is accomplished. This is expected behavior. But when A calls B, and B parks A, this will be current behavior: After B parks A, B is hung up by the system, and the 'h' (hangup) exten gets run, but the channel mentioned will be a derivative of A's... Thus, if A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be those relating to Channel A. And, in the case where A is reconnected to B after the park time expires, when both parties hang up after the joyful reunion, no h-exten will be run at all. In the case where C picks up A from the parking lot, when either A or C hang up, the h-exten will be run for the C channel. CDR's are a separate issue, and not addressed here. As to WHY this strange behavior occurs, the answer lies in the procedure followed to accomplish handing over the channel to the parking manager thread. This procedure is called masquerading. In the process, a duplicate copy of the channel is created, and most of the active data is given to the new copy. The original channel gets its name changed to XXX<ZOMBIE> and keeps the PBX information for the sake of the original thread (preserving its role as a call originator, if it had this role to begin with), while the new channel is without this info and becomes a call target (a "peer"). In this case, the parking lot manager thread is handed the new (masqueraded) channel. It will not run an h-exten on the channel if it hangs up while in the parking lot. The h exten will be run on the original channel instead, in the original thread, after the bridge completes. See bug 13820 for our intentions as to how to clean up the h exten behavior. Review: http://reviewboard.digium.com/r/29/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	21012610d2	Merged revisions 165890 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r165890 \| russell \| 2008-12-19 09:05:09 -0600 (Fri, 19 Dec 2008) \| 17 lines Merged revisions 165889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165889 \| russell \| 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008) \| 9 lines Ensure that the chanspy datastore is fully initialized. This patch resolved some random crash issues observed by a user on a BSD system (closes issue #14111) Reported by: ys Patches: app_chanspy.c.diff uploaded by ys (license 281) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@165891 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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