3826 Commits (5b8a1e8bc54f70f8651e1d2d60ee8317eed823c2)

Author	SHA1	Message	Date
Tilghman Lesher	461c3de2ed	Merged revisions 297713 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297713 | tilghman | 2010-12-06 18:21:50 -0600 (Mon, 06 Dec 2010) | 15 lines Merged revisions 297689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010) | 8 lines Don't create a Local channel if the target extension does not exist. (closes issue #18126) Reported by: junky Patches: followme.diff uploaded by junky (license 177) (partially restructured by me to avoid a possible memory leak) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@297733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	3433890c9a	Merged revisions 297229 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297229 | russell | 2010-12-02 07:16:47 -0600 (Thu, 02 Dec 2010) | 13 lines Merged revisions 297228 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010) | 6 lines Add "DAHDI" to a couple of app_meetme error messages. This is in response to some questions on IRC. To the user, there was nothing that made it obvious that this error had anything to do with DAHDI not being loaded. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@297245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	38b81d2772	Merged revisions 296869 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296869 | jpeeler | 2010-11-30 18:24:58 -0600 (Tue, 30 Nov 2010) | 11 lines Merged revisions 296868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010) | 4 lines Properly restore backup information file when hanging up during message prepending. ABE-2654 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@296870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	82ee0bc14e	DOC: Conference number can be omitted; if omitted, all users in a meetme are listed. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@296787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	b4f92dec2c	Merged revisions 296466 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r296466 | tilghman | 2010-11-27 04:39:01 -0600 (Sat, 27 Nov 2010) | 5 lines 18 characters is too short for most date/times (20 is the usual, but we add more in case of greater precision). (closes issue #18369) Reported by: tnakonz ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@296467 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	30a7e71c27	Merged revisions 296001 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296001 | russell | 2010-11-24 11:03:16 -0600 (Wed, 24 Nov 2010) | 45 lines Merged revisions 296000 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines Handle failures building translation paths more effectively. The problem scenario occurred on a heavily loaded system that was using the codec_dahdi module and exceeded the hardware transcoding capacity. The failure mode at that point was not good. The report came in to us as an Asterisk lock-up. The "core show locks" shows a ton of threads locked up (but no obvious deadlock). Upon deeper investigation, when the system is in this state, the CPU was maxed out. The CPU was being consumed by the Asterisk logger spewing messages on every audio frame for calls set up after transcoder capacity was reached. The purpose of this patch is to make Asterisk handle failures to create a translation path in a more graceful manner. If we can't translate, then the call just needs to be dropped, as it's not going to work. These are the changes: 1) In set_format() of channel.c (which is called by set_read_format() and set_write_format()), it was ignoring if ast_translator_build_path() failed and returned NULL. It now pays attention to that case and returns a result reflecting failure. With this change in place, the bridging code will immediately detect a failure and end the bridge instead of proceeding to try to bridge frames that can't be translated and making channel drivers freak out by sending them frames in a format they weren't expecting. 2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was ignored. It is now reflected in the return value of the function. This didn't turn out to have any affect on the bug, but seemed like a good change to leave in. 3) In app_dial(), when only sending a call to a single endpoint, it will attempt to do some bridging of its own of early audio. It uses make_compatible() when it's going to do this. However, it ignored failure from make compatible. So, even with the fix from #1, if there was early audio going through app_dial, there would still be a period of invalid frames passing through. After detecting failure here, Dial() exits. ABE-2658 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@296002 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	c08103f033	Merged revisions 295843 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295843 | rmudgett | 2010-11-22 13:28:23 -0600 (Mon, 22 Nov 2010) | 53 lines Merged revisions 295790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. To recreate the problem: 1) Party A calls Party B 2) Invoke CLI "channel redirect" command to redirect channel call leg associated with A. 3) All associated channels are hung up. Note that if the CLI command were done on the channel call leg associated with B it works. This regression was a result of the fix for issue #16946 (https://reviewboard.asterisk.org/r/740/). The regression affects all features that use an async goto to execute the dialplan because of an external event: Channel redirect, AMI redirect, SIP REFER, and FAX detection. The struct ast_channel._softhangup code is a mess. The variable is used for several purposes that do not necessarily result in the call being hung up. I have added doxygen comments to describe how the various _softhangup bits are used. I have corrected all the places where the variable was tested in a non-bit oriented manner. The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so the soft hangup requests that do not normally result in a hangup do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171) Reported by: SantaFox (closes issue #18185) Reported by: kwemheuer (closes issue #18211) Reported by: zahir_koradia (closes issue #18230) Reported by: vmarrone (closes issue #18299) Reported by: mbrevda (closes issue #18322) Reported by: nerbos Review: https://reviewboard.asterisk.org/r/1013/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	ddb80391f6	Patch for deadlock from ordering issue between channel/queue locks in app_queue ... (set_queue_variables). (closes issue #18031) Reported by: rain Review: https://reviewboard.asterisk.org/r/1018/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	f1abd401b9	Merged revisions 294910 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294910 | jpeeler | 2010-11-12 15:14:23 -0600 (Fri, 12 Nov 2010) | 4 lines Return correct error code if lock path fails. The recent changes to open_mailbox actually caused it to be fixed, but let's be consistent. Reported by alecdavis in asterisk-dev. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294911 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	06ac20454e	Merged revisions 294904 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294904 | jpeeler | 2010-11-12 14:51:15 -0600 (Fri, 12 Nov 2010) | 23 lines Merged revisions 294903 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010) | 16 lines Fix regression causing abort in voicemail after opening a mailbox with no mesgs. In order to be more safe, some error handling code was changed to respect more error conditions including the potential memory allocation failure for deleted and heard message tracking introduced in 293004. However, last_message_index returns -1 for zero messages (perhaps as expected) and was triggering the stricter error checking. Because last_message_index is only called directly in one place, just return 0 from open_mailbox (for file based storage) when no messages are detected unless a real error has occurred. (closes issue #18240) Reported by: leobrown Patches: bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585) Tested by: pabelanger ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	6cbda6ed92	Merged revisions 293118 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293118 | jpeeler | 2010-10-26 13:33:24 -0500 (Tue, 26 Oct 2010) | 36 lines Merged revisions 293004 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010) | 29 lines Fix inprocess_container in voicemail to correctly restrict max messages. The comparison function logic was off, so the number of sessions for a given mailbox were not being incremented properly. This problem caused the maximum number of messages per folder to not be respected when simultaneously leaving multiple voicemails just below the threshold. These problems should be fixed by the above, but just in case: Fixed resequence_mailbox to rely on the actual number of detected number of files in a directory rather than just assuming only 10 messages more than the maximum had been left. Also if more messages than the maximum are deleted they are actually removed now. The second purpose of this commit should have been separated out probably, but is related to the above. Again, if the number of messages in a given voicemail folder exceeds the maximum set limit make sure to allocate enough space for the deleted and heard index tracking array. A few random fixes: There was a forgotten decrement of the inprocess count in imap_store_file. When using IMAP storage, do not look in the directory where file based storage messages may still reside and influence the message count. Ensure to use only the first format in sendmail. ABE-2516 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	83ed33746e	Application not properly unregister in voicemail ... (closes issue #18128) Reported by: junky Patches: vm_unregister.diff uploaded by junky (license 177) Tested by: pabelanger, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	d4a74bd2ae	Merged revisions 292412 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292412 | pabelanger | 2010-10-20 20:05:45 -0400 (Wed, 20 Oct 2010) | 17 lines Merged revisions 292411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct 2010) | 10 lines Record priv-recordintro as sln, not gsm This removes the gsm->sln step when transcoding priv-recordintro. (closes issue #18176) Reported by: pabelanger Patches: chan_sip.diff uploaded by pabelanger (license 224) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	aecdf5d980	Merged revisions 292226 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292226 | jpeeler | 2010-10-18 16:54:38 -0500 (Mon, 18 Oct 2010) | 18 lines Merged revisions 292223 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010) | 11 lines Fix improper operator key acceptance and clean up temp recording files. This is a fix for when pressing the operator key after recording an unavailable, busy, name, or temporary message in mailbox options. The operator key should not be accepted here, but should be allowed during the message recording. If the operator key is pressed during ensure the file is saved or deleted as apporopriate. Also, ensure removal of temporary recorded files after an early hang up or when message acceptance confirmation times out. ABE-2518 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	966c392632	Merged revision 290613 from ... https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r290613 | rmudgett | 2010-10-06 13:42:41 -0500 (Wed, 06 Oct 2010) | 5 lines Eliminate a redundant test for AST_CONTROL_REDIRECTING. Eliminate redundant test for AST_CONTROL_REDIRECTING that prevents running the redirecting interception macro if it is defined. .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290614 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	2a2b6a96d4	Merged revisions 290375 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r290375 | dvossel | 2010-10-05 14:54:50 -0500 (Tue, 05 Oct 2010) | 10 lines Fixes PickupChan() not working with full channel name. (closes issue #18011) Reported by: schern Patches: app_directed_pickup.c.2.patch uploaded by schern (license 995) app_directed_pickup.c.trunk.patch uploaded by schern (license 995) Tested by: schern, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290376 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	9b6af22c3d	Merged revisions 289874 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289874 | tilghman | 2010-10-01 23:45:49 -0500 (Fri, 01 Oct 2010) | 15 lines Merged revisions 289873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010) | 8 lines When forwarding a message, a prepend means that the filesystem will always have a better copy. (closes issue #17803) Reported by: dpetersen Patches: 20100923__issue17803.diff.txt uploaded by tilghman (license 14) Tested by: dpetersen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@289875 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	37ad96f682	Merged revisions 289425 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289425 | russell | 2010-09-30 10:37:29 -0500 (Thu, 30 Sep 2010) | 15 lines Merged revisions 289424 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010) | 8 lines Fix a crash in app_sms. Since the data being passed to the generator callback is on the stack of the SMS() application, we must ensure that the generator is stopped before the application exits. ABE-2587 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@289426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e00c4dcc6d	Solaris compatibility fixes ... Review: https://reviewboard.asterisk.org/r/942/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@289104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	eca6952995	Simplify locking code for REDIRECTING interception macro when forwarding a call. ... Simplified the locking code by using a local copy of the redirecting party information in app_dial.c:do_forward() and app_queue.c:wait_for_answer() for launching the REDIRECTING interception macro when a call is forwarded. Reduced the lock time of the 'o->chan' and 'in' channels. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@288080 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Brett Bryant	71cbbd60de	Merged revisions 287759 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r287759 | bbryant | 2010-09-20 19:58:26 -0400 (Mon, 20 Sep 2010) | 23 lines Merged revisions 287758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010) | 16 lines Fix misvalidation of meetme pins in conjunction with the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a user and admin pin setup for your conference, using the user pin would gain you admin priviledges. Also, when no user pin was set, an admin pin was, the 'a' MeetMe flag wasn't used, and the user tried to enter a conference then they were still prompted for a pin and forced to hit #. (closes issue #17908) Reported by: kuj Patches: pins_2.patch uploaded by kuj (license 1111) Tested by: kuj Review: [full review board URL with trailing slash] ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	72718e1183	Merged revisions 287387 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r287387 | tilghman | 2010-09-17 16:08:00 -0500 (Fri, 17 Sep 2010) | 14 lines Merged revisions 287386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010) | 7 lines Blank columns should get set on reload, not ignored. (closes issue #16893) Reported by: haakon Patches: 20100818__issue16893.diff.txt uploaded by tilghman (license 14) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	daf14509c5	Set the default for "autofill" and "shared_lastcall" to "yes" in queues.conf. ... Review: https://reviewboard.asterisk.org/r/922/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287193 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	149f98f25b	Merged revisions 286998 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286998 | jpeeler | 2010-09-15 15:28:02 -0500 (Wed, 15 Sep 2010) | 14 lines Merged revisions 286941 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010) | 7 lines Ensure mailbox is not filled to capacity before doing message forwarding. Specifically, before prompting to record a prepended message the capacity is checked first. If the mailbox is full the extension will be reprompted. ABE-2517 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Brett Bryant	69e4f34914	Merged revisions 285532 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285532 | bbryant | 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010) | 8 lines Fixes a bug with MeetMe where after announcing the amount of time left in a conference, if music on hold was playing, it doesn't restart. (closes issue #17408) Reported by: sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by sysreq (license 1009) Tested by: sysreq ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Brett Bryant	8131d12a71	Merged revisions 285196 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285196 | bbryant | 2010-09-07 13:49:07 -0400 (Tue, 07 Sep 2010) | 17 lines Merged revisions 285194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07 Sep 2010) | 10 lines Fixes voicemail.conf issues where mailboxes with passwords that don't precede a comma would throw unnecessary error messages. (closes issue #15726) Reported by: 298 Patches: M15726.diff uploaded by junky (license 177) Tested by: junky Review: [full review board URL with trailing slash] ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	2e701efc81	Merged revisions 284897 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284897 | twilson | 2010-09-03 11:20:45 -0500 (Fri, 03 Sep 2010) | 12 lines Merged revisions 284881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010) | 5 lines Properly detect when a sound file doesn't exist ast_fileexists returns -1 for error and 0 for a non-existant file. The existing code treated missing files as though they existed. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284921 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8bc90ad39a	Merged revisions 284631 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r284631 | tilghman | 2010-09-02 00:30:16 -0500 (Thu, 02 Sep 2010) | 7 lines Don't reset queue stats on a module reload. (closes issue #17535) Reported by: raarts Patches: 20100819__issue17535.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284632 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	7e3f95e00a	When optional_api is non-optional, force dependent modules to be loaded. ... (closes issue #17707) Reported by: ira Patches: 20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/876/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284610 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	96f182bec0	Merged revisions 284280 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r284280 | tilghman | 2010-08-30 17:27:06 -0500 (Mon, 30 Aug 2010) | 11 lines Fix 3 coding errors: 1) After we close FD, we should not be trying to write to it. 2) Call _exit(0), not exit(0), to avoid running shutdown routines in a child. 3) Use endian, not processor, detection to ensure bytes are written in the correct order. (closes issue #15706) Reported by: modelnine Patches: asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine (license 865) Tested by: gmartinez ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284281 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	c3ad0f569d	Add an argument missing from the CELGenUserEvent documentation. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282979 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	ddceaeeae9	Merged revisions 281722 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r281722 | tilghman | 2010-08-11 10:17:20 -0500 (Wed, 11 Aug 2010) | 7 lines Only set status TIMEOUT, if we have no digits. (closes issue #15188) Reported by: jcovert Patches: app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license 551) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281723 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	325a4cd89a	Merged revisions 281567 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r281567 | russell | 2010-08-10 12:47:13 -0500 (Tue, 10 Aug 2010) | 15 lines Merged revisions 281566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010) | 8 lines Reset visible indication after answer. (closes issue #17641) Reported by: klaus3000 Patches: ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by klaus3000 (license 65) Tested by: schmidts ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281568 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
TransNexus OSP Development	cc67d321e8	Fixed the issue caused by EXTEN including user parameters. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281497 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	0fed1acd2a	Merged revisions 280671 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280671 | tilghman | 2010-08-02 16:26:11 -0500 (Mon, 02 Aug 2010) | 2 lines Allow the pipe, but also allow the comma ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jean Galarneau	ae2e66e707	Merged revisions 280345 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r280345 | jeang | 2010-07-29 11:01:35 -0500 (Thu, 29 Jul 2010) | 10 lines Merged revisions 280341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) | 2 lines Fix a dsp structure leak occuring when a local channel is put into a meetme conference, then masquaraded away. ABE-2422 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280346 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	5b52f62450	Merged revisions 280160 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280160 | seanbright | 2010-07-28 12:51:11 -0400 (Wed, 28 Jul 2010) | 8 lines Plug a reference leak in app_queue when adding members dynamically. (closes issue #17738) Reported by: bobwienholt Patches: issue17738.patch uploaded by bobwienholt (license 950) Tested by: bobwienholt, seanbright ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280161 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	6341d1b2ad	Merged revisions 279207 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r279207 | rmudgett | 2010-07-23 17:11:23 -0500 (Fri, 23 Jul 2010) | 14 lines Merged revisions 279206 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) | 7 lines SIP promiscuous redirect could fail to dial the redirect. The ast_channel was created with one variable to ast_request() but the call to ast_call() that initiates the outgoing call was using a different variable. The two variables are not equivalent if the call_forward string included a channel technology specifier. e.g., SIP/200 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	9bb8dc67e7	Ensure realtime conferences are treated the same as static conferences when trying to find an empty one. ... Also, parse the useropts properly, when retrieving from realtime, and add them to the existing flags. (closes issue #17502) Reported by: kenji Patches: 20100720__issue17502.diff.txt uploaded by tilghman (license 14) Tested by: kenji git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278463 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	ebf651105e	Merged revisions 278261 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20 Jul 2010) | 7 lines Delete IMAP messages in reverse order, to ensure reordering after each expunge does not cause deletion of the wrong message. (closes issue #16350) Reported by: noahisaac Patches: 20100623__issue16350.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278275 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	b4e18d5660	Add load priority order, such that preload becomes unnecessary in most cases ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	5b8a8fc6c8	Fix reporting estimated queue hold time. ... Just say the number of seconds (after minutes) rather than doing some incorrect calculation with respect to minutes. (closes issue #17498) Reported by: corruptor Patches: holdesecs_bug.diff uploaded by corruptor (license 253) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	b73c1377e5	Add missing handling for ringing state for use with queue empty options. ... (closes issue #17471) Reported by: jazzy Patches: app_queue.c.diff uploaded by jazzy (license 1056) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277366 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	8eb9e0b938	Merged revisions 277182 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul 2010) | 8 lines Total analysis time error with SIP and silence suppression When using app_amd with SIP providers that have silence suppression on, the iTotalTime count increases exponentially. (closes issue #17656) Reported by: juls ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277183 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	65203b12dd	Add a dialplan function to check if a queue exists: QUEUE_EXISTS ... Review: https://reviewboard.asterisk.org/r/777/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	cf7bbcc4c6	Expand the caller ANI field to an ast_party_id ... Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	ec37ffbdaf	ast_callerid restructuring ... The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	6535a1d0ed	Merged revisions 275773 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010) | 12 lines Make user removals and traversals thread safe in meetme. Race conditions present in meetme involving the user list where a lack of locking has the potential for a user to be removed during a traversal or as in the case of the reporter after checking if the list is empty could cause a crash. Fixing this was done by convering the userlist to an ao2 container. (closes issue #17390) Reported by: Vince Review: https://reviewboard.asterisk.org/r/746/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276074 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
TransNexus OSP Development	f1df8ea2bf	Added support for indirect work mode. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Eliel C. Sardanons	7eafb1a763	When creating a conference for a unit test, it is not mandatory to open a ... dahdi pseudo channel, so if we fail doing it, continue creating the conference. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago