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2011-02-16Update .version file, ChangLog, removed old summary files, and merged in ↵lmadsen1-1/+0
change to app_queue. git-svn-id: http://svn.digium.com/svn/asterisk/tags/ f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-08Indicate log level argument for Log() is not optionalpabelanger1-1/+1
(closes issue #18586) Reported by: kshumard Patches: app_verbose.c.patch uploaded by kshumard (license 92) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@301176 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-07Initialize useropts/adminopts in case there is no column in the realtime DB.qwell1-2/+2
(closes issue #18182) Reported by: dimas Patches: v1-18182.patch uploaded by dimas (license 88) Tested by: dimas git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@301089 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-07Fix regression causing forwarding voicemails to not work with file storage.jpeeler1-6/+14
I had actually already fixed this in 295200 in 1.4 and thought it wasn't missing in the other branches for some reason. (closes issue #18358) Reported by: cabal95 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@301046 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-07Merged revisions 300918 via svnmerge from jpeeler1-4/+9
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011) | 7 lines Ensure good bye prompt in voicemail is played at the correct time. Specifically in the case of timing out but not leaving voicemail nothing should be heard. And when leaving voicemail it should be heard. ABE-2647 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@300951 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-28Documentation typopabelanger1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@299864 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-16Merged revisions 298683 via svnmerge from jpeeler1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16 Dec 2010) | 2 lines After recording only silence for a voicemail prepending, restore backup files. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@298684 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-16Merged revisions 298596 via svnmerge from jpeeler1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010) | 7 lines Fix improper hangup when doing an attended transfer to queue. Had to indicate ringing in wait_for_answer so the attended transfer code would not try and hang up the local channel it created, which would kill the call. ABE-2624 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@298597 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-07Merged revisions 297689 via svnmerge from tilghman1-2/+8
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@297713 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-02Merged revisions 297228 via svnmerge from russell1-2/+2
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@297229 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Merged revisions 296868 via svnmerge from jpeeler1-0/+10
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@296869 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-2718 characters is too short for most date/times (20 is the usual, but we add ↵tilghman1-2/+2
more in case of greater precision). (closes issue #18369) Reported by: tnakonz git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296466 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Merged revisions 296000 via svnmerge from russell1-1/+10
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@296001 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-22Merged revisions 295790 via svnmerge from rmudgett1-11/+11
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@295843 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-12Return correct error code if lock path fails. The recent changes to ↵jpeeler1-1/+1
open_mailbox actually caused it to be fixed, but let's be consistent. Reported by alecdavis in asterisk-dev. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@294910 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-12Merged revisions 294903 via svnmerge from jpeeler1-1/+2
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@294904 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-26Merged revisions 293004 via svnmerge from jpeeler1-21/+91
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@293118 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-21Merged revisions 292411 via svnmerge from pabelanger1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@292412 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-18Merged revisions 292223 via svnmerge from jpeeler1-2/+12
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@292226 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-05Fixes PickupChan() not working with full channel name.dvossel1-8/+16
(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: http://svn.digium.com/svn/asterisk/branches/1.6.2@290375 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-02Merged revisions 289873 via svnmerge from tilghman1-12/+6
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@289874 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-30Merged revisions 289424 via svnmerge from russell1-0/+6
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@289425 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-20Merged revisions 287758 via svnmerge from bbryant1-6/+12
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@287759 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-17Merged revisions 287386 via svnmerge from tilghman1-4/+4
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@287387 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-15Merged revisions 286941 via svnmerge from jpeeler1-5/+29
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@286998 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-09Fixes an issue with dialplan pattern matching where the specificity for ↵bbryant7-14/+23
pattern ranges and pattern special characters was inconsistent. (closes issue #16903) Reported by: Nick_Lewis Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@285710 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-02Allow the pipe, but also allow the commatilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280671 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29Merged revisions 280341 via svnmerge from jeang1-0/+3
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@280345 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-28Plug a reference leak in app_queue when adding members dynamically.seanbright1-0/+1
(closes issue #17738) Reported by: bobwienholt Patches: issue17738.patch uploaded by bobwienholt (license 950) Tested by: bobwienholt, seanbright git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280160 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Merged revisions 279206 via svnmerge from rmudgett2-6/+12
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@279207 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-21Merged revisions 278463 via svnmerge from tilghman1-7/+52
https://origsvn.digium.com/svn/asterisk/trunk ........ r278463 | tilghman | 2010-07-21 10:56:05 -0500 (Wed, 21 Jul 2010) | 11 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.2@278464 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Merged revisions 278275 via svnmerge from tilghman1-41/+49
https://origsvn.digium.com/svn/asterisk/trunk ................ r278275 | tilghman | 2010-07-20 17:40:19 -0500 (Tue, 20 Jul 2010) | 14 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.2@278276 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Merged revisions 277488 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r277488 | jpeeler | 2010-07-16 16:16:08 -0500 (Fri, 16 Jul 2010) | 10 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.2@277489 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Merged revisions 277366 via svnmerge from jpeeler1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r277366 | jpeeler | 2010-07-16 14:22:49 -0500 (Fri, 16 Jul 2010) | 7 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.2@277377 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Merged revisions 277183 via svnmerge from pabelanger1-3/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r277183 | pabelanger | 2010-07-16 13:13:46 -0400 (Fri, 16 Jul 2010) | 15 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.2@277188 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-13Merged revisions 276074 via svnmerge from jpeeler1-109/+194
https://origsvn.digium.com/svn/asterisk/trunk ................ r276074 | jpeeler | 2010-07-13 12:37:40 -0500 (Tue, 13 Jul 2010) | 19 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.2@276121 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Merged revisions 275028 via svnmerge from mnicholson1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r275028 | mnicholson | 2010-07-09 11:05:58 -0500 (Fri, 09 Jul 2010) | 15 lines Merged revisions 275027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul 2010) | 8 lines Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx via the G option in app_dial (closes issue #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff uploaded by mnicholson (license 96) Tested by: jamicque, mnicholson ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@275029 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-02Merged revisions 273714 via svnmerge from tilghman1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r273714 | tilghman | 2010-07-02 11:57:28 -0500 (Fri, 02 Jul 2010) | 2 lines The switch fallthrough could create some errorneous situations, so best to force directly to the default case. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273716 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-02Fix typos reported by Lintiantzafrir1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273642 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-01Merged revisions 273522 via svnmerge from jpeeler1-2/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r273522 | jpeeler | 2010-07-01 15:28:15 -0500 (Thu, 01 Jul 2010) | 21 lines Merged revisions 273474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010) | 14 lines Allow admin user to join conference without using admin mode and no user pin. Configuring the conference in meetme.conf like the following: conf => 2345,,6666 did not prompt for pin when used without admin mode. This meant that the conference could not be joined as an admin even if the user knew the correct pin. The original bug report was submitted claiming that the blank user pin should deny entry into the conference. I think a better way to handle this would be with a feature enhancement that used the following syntax: conf => 2345,X,6666 - where X denotes no acceptable pin allowed (closes issue #15704) Reported by: modelnine ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273529 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-01Merged revisions 273355 via svnmerge from jpeeler1-0/+12
https://origsvn.digium.com/svn/asterisk/trunk ................ r273355 | jpeeler | 2010-07-01 10:12:31 -0500 (Thu, 01 Jul 2010) | 19 lines Merged revisions 273354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010) | 12 lines Ensure channel placed in meetme in ringing state is properly hung up. An outgoing channel placed in meetme while still ringing which was then hung up would not exit meetme and the channel was not properly destroyed. Specifically checking for this scenario by looking at the appropriate control frames resolves the issue. (closes issue #15871) Reported by: Ivan Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan (license 229) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@273356 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Merged revisions 272368 via svnmerge from mnicholson1-1/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r272368 | mnicholson | 2010-06-23 17:36:49 -0500 (Wed, 23 Jun 2010) | 16 lines Merged revisions 272367 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 This version of the patch only adds AgentComplete for attended transfers. It was already present for blind transfers. ........ r272367 | mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8 lines Send AgentComplete manager events in the event of blind and attended transfers. (closes issue #16819) Reported by: elbriga Patches: app_queue.diff uploaded by elbriga (license 482) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@272369 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Revert previous commit, ast_test_flag64 does not exist in 1.6.2pabelanger1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@272263 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Merged revisions 272259 via svnmerge from pabelanger1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r272259 | pabelanger | 2010-06-23 17:06:15 -0400 (Wed, 23 Jun 2010) | 2 lines Fix previous merge. ast_test_flag != ast_test_flag64 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@272261 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Merged revisions 272257 via svnmerge from pabelanger1-3/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r272257 | pabelanger | 2010-06-23 17:00:00 -0400 (Wed, 23 Jun 2010) | 19 lines Merged revisions 272255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun 2010) | 12 lines First caller into a dynamic conference now enter pin once. If MeetMe is configured to use dynamic conference numbers, then the first caller (which creates the conference) had to enter the PIN number twice. (closes issue #15878) Reported by: shawkris Patches: issue15878.patch uploaded by pabelanger (license 224) Tested by: pabelanger ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@272258 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Merged revisions 272146 via svnmerge from twilson1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r272146 | twilson | 2010-06-23 13:39:20 -0500 (Wed, 23 Jun 2010) | 2 lines Don't start the sla thread unless we realy need it ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@272149 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Merged revisions 272109 via svnmerge from twilson1-2/+19
https://origsvn.digium.com/svn/asterisk/trunk ........ r272109 | twilson | 2010-06-23 12:21:40 -0500 (Wed, 23 Jun 2010) | 12 lines Make sure reload updates SLA config Even if there are no stations or trunks defined, we need to start the sla thread to make sure we get the reload event. Also, when doing a reload we need to remove the existing trunks and stations or they end up hanging around. (closes issue #16818) Reported by: mbonin Patches: sla_reload.patch uploaded by twilson (license 396) Tested by: twilson ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@272124 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-17Merged revisions 271089 via svnmerge from pabelanger1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r271089 | pabelanger | 2010-06-16 20:30:51 -0400 (Wed, 16 Jun 2010) | 5 lines option w[(secs)] incorrectly capitalized in xmldoc (closes issue #17516) Reported by: karlfife ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@271090 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-08Merged revisions 269083 via svnmerge from mnicholson1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r269083 | mnicholson | 2010-06-08 13:50:45 -0500 (Tue, 08 Jun 2010) | 9 lines Don't pass null to manager_event() (closes issue #17087) Reported by: bklang Patches: app-fax-null-sprintf1.diff uploaded by mnicholson (license 96) Tested by: bklang ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@269084 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-25Merged revisions 265611 via svnmerge from mnicholson1-7/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r265611 | mnicholson | 2010-05-25 12:00:11 -0500 (Tue, 25 May 2010) | 15 lines Merged revisions 265610 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May 2010) | 8 lines Don't mark the cdr records of unanswered queue calls with "NOANSWER". This restores the behavior prior to r258670. (closes issue #17334) Reported by: jvandal Patches: queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested by: aragon, jvandal ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@265612 f38db490-d61c-443f-a65b-d21fe96a405b