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2011-04-21New HD ConfBridge conferencing application.David Vossel
Includes a new highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8khz-192khz. Review: https://reviewboard.asterisk.org/r/1147/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-19Merged revisions 314203 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314203 | lmadsen | 2011-04-19 09:24:25 -0500 (Tue, 19 Apr 2011) | 15 lines Merged revisions 314202 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r314202 | lmadsen | 2011-04-19 09:23:39 -0500 (Tue, 19 Apr 2011) | 7 lines Update seconds to milliseconds in ast_verb output. (closes issue #19084) Reported by: smurfix Patches: app_dial.patch uploaded by smurfix (license 547) Tested by: lmadsen, smurfix ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314204 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-19Add explanation of strange flag setup in app_meetme (stolen from Mark's ↵Olle Johansson
message to asterisk-dev) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-18Merged revisions 314068 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314068 | rmudgett | 2011-04-18 11:02:12 -0500 (Mon, 18 Apr 2011) | 7 lines Unclear code in app_dial.c. Make code formatting clear. (closes issue #19134) Reported by: oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-13Merged revisions 313517 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313517 | rmudgett | 2011-04-12 17:35:53 -0500 (Tue, 12 Apr 2011) | 12 lines Bring the dumpchan application inline with "core show channel". * Added fields that are in "core show channel" to dumpchan output. * Fixed reuse of formatbuf before the previous string stored there was used by snprintf. All output strings now have their own buffer. * Adjusted the buffer sizes to not be so abusive of the stack now that there are more buffers. Change requested by oej. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-11Merged revisions 313368-313369 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313368 | rmudgett | 2011-04-11 18:03:02 -0500 (Mon, 11 Apr 2011) | 2 lines Backport a restructuring change from trunk to make the next change stand out. ........ r313369 | rmudgett | 2011-04-11 18:08:02 -0500 (Mon, 11 Apr 2011) | 13 lines Frames from the inbound channel should go to all outbound channels in app_dial.c. In app_dial.c:wait_for_answer() frames from the inbound channel should be sent to all outbound channels instead of only if there is just one outbound channel. Control frames like AST_CONTROL_CONNECTED_LINE need to be passed to all of the the outbound channels. This can happen if a blond transfer is done by a remote switch on the inbound channel. JIRA AST-443 JIRA SWP-2730 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313383 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-07app_voicemail: close_mailbox change LOG_WARNING to LOG_NOTICEAlec L Davis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-05Minor change to 'L' option for meetme to include some verb statements for ↵Jonathan Rose
the option. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-01Merged revisions 312211 via svnmerge from Alec L Davis
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312211 | alecdavis | 2011-04-01 22:03:11 +1300 (Fri, 01 Apr 2011) | 36 lines Merged revisions 312210 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312210 | alecdavis | 2011-04-01 21:47:29 +1300 (Fri, 01 Apr 2011) | 29 lines Merged revisions 312174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312174 | alecdavis | 2011-04-01 21:29:49 +1300 (Fri, 01 Apr 2011) | 23 lines voicemail: get real last_message_index and count_messages, ODBC resequence change last_message_index to read the max msgnum stored in the database change count_messages to actually count the number of messages. last_message_index change: This fixed overwriting of the last message if msgnum=0 was missing. Previously every incoming message would overwrite msgnum=1. count_messages change: allows us to detect when requencing is required in opneA_mailbox. resequence enabled for ODBC storage: Assists with fixing up corrupt databases with gaps, but only when a user actively opens there mailboxes. (closes issue #18692,#18582,#19032) Reported by: elguero Patches: based on odbc_resequence_mailbox2.1.diff uploaded by elguero (license 37) Tested by: elguero, nivek, alecdavis Review: https://reviewboard.asterisk.org/r/1153/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-01Merged revisions 312117 via svnmerge from Alec L Davis
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312117 | alecdavis | 2011-04-01 20:32:12 +1300 (Fri, 01 Apr 2011) | 29 lines Merged revisions 312103 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312103 | alecdavis | 2011-04-01 20:25:54 +1300 (Fri, 01 Apr 2011) | 22 lines Merged revisions 312070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312070 | alecdavis | 2011-04-01 19:46:56 +1300 (Fri, 01 Apr 2011) | 16 lines app_voicemail: close_mailbox needs to respect additional messages while mailbox is open. close_mailbox leave gaps in message sequence if messages are deleted and new messages arrive during this time, this is because the shuffle down to slot 0, only shuffles the number of pre-existing messages when mailbox is opened, ignoring new arrivals. Fix: in close_mailbox re-evaluate number of messages before the shuffle, this then includes new arrivals. Happens on filebased or ODBC storage. (issues #19032,#18582,#18692,#18998) Reported by: alecdavis,tootai,afosorio Review: https://reviewboard.asterisk.org/r/1153/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-28Merged revisions 311751 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311751 | russell | 2011-03-28 17:00:01 -0500 (Mon, 28 Mar 2011) | 2 lines Cross-reference VoiceMail() and VoiceMailMain() in the xml docs. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311752 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-23Merged revisions 311615 via svnmerge from Brett Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311615 | bbryant | 2011-03-23 17:54:11 -0400 (Wed, 23 Mar 2011) | 8 lines This patch fixes a bug with MeetMe behavior where the 'P' option for always prompting for a pin is ignored for the first caller. (closes issue #18070) Reported by: mav3rick Review: https://reviewboard.asterisk.org/r/1132/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-22Merged revisions 311497 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r311497 | dvossel | 2011-03-22 10:25:24 -0500 (Tue, 22 Mar 2011) | 9 lines Merged revisions 311496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r311496 | dvossel | 2011-03-22 10:24:45 -0500 (Tue, 22 Mar 2011) | 2 lines Fixes memory leak in MeetMe AMI action ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-18Adds an option to FollowMe that isn't useful for the bug it was made to ↵Jonathan Rose
solve. Still, due to the nature of FollowMe, it makes sense to have this option since it keeps apps bound to channels that would otherwise go away from being lost. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-18Merged revisions 311295 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r311295 | rmudgett | 2011-03-17 21:22:07 -0500 (Thu, 17 Mar 2011) | 35 lines Merged revision 310986 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed, 16 Mar 2011) | 28 lines Dial() o option broke when connected line feature added. The patch restores the o option behavior and adds the ability to specify the CallerID. The Dial o and f options are complementary to each other. The o option stores the CallerID on the outgoing channel as the channel's CallerID. The f option forces the CallerID sent by the outgoing channel. o(x) - The argument 'x' is optional. If not present, then specify that the CallerID that was present on the *calling* channel be stored as the CallerID on the *called* channel. This was the behavior of Asterisk 1.0 and earlier. If present, then specify the CallerID stored on the *called* channel. Note that o(${CALLERID(all)}) is similar to option o without parameters. f(x) - The argument 'x' is optional and its presence changes the behavior of this option. If not present, then force the outgoing CallerID on a call-forward or deflection to the dialplan extension for this Dial() using a dialplan 'hint'. For example, some PSTNs do not allow CallerID to be set to anything other than the numbers assigned to you. If present, then force the outgoing CallerID to 'x'. Patches: jira_abe_2752_dial_fo_options.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA ABE-2752 JIRA SWP-3096 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-17Merged revisions 311197 via svnmerge from Jonathan Rose
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311197 | jrose | 2011-03-17 14:03:34 -0500 (Thu, 17 Mar 2011) | 11 lines This fixes a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call. In addition to the above, it makes certain channel destruction occurs so that applications don't get stuck waiting for datastore destruction while monitored by chanspy. (closes issue #18742) Reported by: jkister Tested by: jkister, jcovert, jrose Review: http://reviewboard.digium.internal/r/106/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-11Mix Monitor: Now with r and t options.Jonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-10Merged revisions 310142 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310142 | tilghman | 2011-03-09 23:53:29 -0600 (Wed, 09 Mar 2011) | 19 lines Merged revisions 310141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310141 | tilghman | 2011-03-09 23:51:37 -0600 (Wed, 09 Mar 2011) | 12 lines Merged revisions 310140 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310140 | tilghman | 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011) | 5 lines Initialize column size to 0 to deal with a potential UnixODBC bug on 64-bit systems. (closes issue #18295) Reported by: pruiz ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-07Merged revisions 309858 via svnmerge from Jonathan Rose
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309858 | jrose | 2011-03-07 16:07:25 -0600 (Mon, 07 Mar 2011) | 22 lines Merged revisions 309857 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r309857 | jrose | 2011-03-07 16:04:44 -0600 (Mon, 07 Mar 2011) | 15 lines Merged revisions 309856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r309856 | jrose | 2011-03-07 16:02:12 -0600 (Mon, 07 Mar 2011) | 8 lines Bug fix for MixMonitor involving filenames with '.' not in the extension Closes issue #18391) Reported by: pabelanger Patches:       bugfix.patch uploaded by jrose (license 1225) Tested by: jrose ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-04Merged revisions 309403 via svnmerge from David Ruggles
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309403 | diruggles | 2011-03-03 20:50:44 -0500 (Thu, 03 Mar 2011) | 23 lines Merged revisions 309356 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r309356 | diruggles | 2011-03-03 19:42:28 -0500 (Thu, 03 Mar 2011) | 16 lines Merged revisions 309355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar 2011) | 9 lines fix small memory leak fix small memory leak caused by a string allocation that wasn't freed (closes issue #18907) Reported by: andy11 Patches: asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 (license 1224) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd ↵David Vossel
audio ConfBridge, and other stuff -Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-15Merged revisions 308010 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308010 | qwell | 2011-02-15 17:34:03 -0600 (Tue, 15 Feb 2011) | 24 lines Merged revisions 308007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308007 | qwell | 2011-02-15 17:33:24 -0600 (Tue, 15 Feb 2011) | 17 lines Merged revisions 308002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) | 10 lines Fix regression that changed behavior of queues when ringing a queue member. This reverts r298596, which was to fix a highly bizarre and contrived issue with a queue member that called into his own queue being transferred back into his own queue. I couldn't reproduce that issue in any way. I think one of the other recent transfer fixes actually fixed this. (closes issue #18747) Reported by: vrban ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-15Merged revisions 307962 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307962 | rmudgett | 2011-02-15 13:52:45 -0600 (Tue, 15 Feb 2011) | 1 line Don't crash when forcing caller id. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-14Merged revisions 307750 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307750 | tilghman | 2011-02-14 00:50:23 -0600 (Mon, 14 Feb 2011) | 23 lines Calling a gosub routine defined in AEL from Dial/Queue ceased to work. A bug in AEL did not distinguish between the "s" extension generated by AEL and an "s" extension that was required to exist by the chan_dahdi (or another channel) that was not supplied with a starting extension. Therefore, AEL made incorrect assumptions about what commands were permissable in the context. This was fixed by making AEL generate a different extension name. However, Dial and Queue make additional assumptions about the name of the default gosub extension. Therefore, they needed to be brought into line with a "macro" rendered by AEL (as a gosub), without breaking traditional dialplans written without the aid of AEL. Related to (issue #18480) Reported by: nivek (closes issue #18729) Reported by: kkm Patches: 20110209__issue18729.diff.txt uploaded by tilghman (license 14) 018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888) Tested by: kkm ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-09Add new manager action MeetmeListRooms.Jeff Peeler
From the submitter: I've added a new manager action to list only the active conferences on an Asterisk system. It shows the same data displayed when you run a 'meetme list' on the Asterisk CLI. (closes issue #17905) Reported by: rcasas Patches: app_meetme.c.patch uploaded by rcasas (license 641) Review: https://reviewboard.asterisk.org/r/874/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08Merged revisions 306967 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306967 | jpeeler | 2011-02-08 13:41:42 -0600 (Tue, 08 Feb 2011) | 16 lines Merged revisions 306966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306966 | jpeeler | 2011-02-08 13:41:21 -0600 (Tue, 08 Feb 2011) | 9 lines Merged revisions 306965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08 Feb 2011) | 1 line fix this line again ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08Merged revisions 306962 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306962 | jpeeler | 2011-02-08 13:25:38 -0600 (Tue, 08 Feb 2011) | 22 lines Merged revisions 306961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306961 | jpeeler | 2011-02-08 13:25:10 -0600 (Tue, 08 Feb 2011) | 15 lines Merged revisions 306960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011) | 9 lines Backup file storing message duration is not used with IMAP_STORAGE, remove code. The message duration is stored in the body of the email when using IMAP_STORAGE, so nothing needs to happen with the backup file. (closes issue #18718) Reported by: kerframil ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-08Merged revisions 306866 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306866 | jpeeler | 2011-02-08 10:21:45 -0600 (Tue, 08 Feb 2011) | 16 lines Merged revisions 306865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306865 | jpeeler | 2011-02-08 10:21:25 -0600 (Tue, 08 Feb 2011) | 9 lines Merged revisions 306864 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08 Feb 2011) | 1 line make this safer and fully correct, pointed out by Steve Davis ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306867 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04Add ISDN display ie text handling options to chan_dahdi.conf.Richard Mudgett
The display ie handling can be controlled independently in the send and receive directions with the following options: * Block display text data. * Use display text in SETUP/CONNECT messages for name. * Use display text for COLP name updates (FACILITY/NOTIFY as appropriate). * Pass arbitrary display text during a call. Sent in INFORMATION messages. Received from any message that the display text was not used as a name. If the display options are not set then the options default to legacy behavior. The arbitrary display text is exchanged between bridged channels using the AST_FRAME_TEXT frame type. To send display text from the dialplan use the SendText() application when the arbitrary display text option is enabled. JIRA SWP-2688 JIRA ABE-2693 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04Merged revisions 306356 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306356 | qwell | 2011-02-04 13:24:29 -0600 (Fri, 04 Feb 2011) | 16 lines Merged revisions 306346 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r306346 | qwell | 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) | 9 lines Don't fallthrough to 'unknown' in the 'ringing' case. This could cause improper exits from the queue. (closes issue #18499) Reported by: zaltar Patches: app_queue.patch uploaded by zaltar (license 1148) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04Merged revisions 306324 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306324 | rmudgett | 2011-02-04 12:53:06 -0600 (Fri, 04 Feb 2011) | 9 lines Don't send redirecting updates to the caller if the dialplan forked the call. Each fork in the dial could be redirected and confuse the caller. For ISDN the DivLeg1 and DivLeg3 messages would get confused because ISDN redirects calls in sequence not in parallel. * Also fixed a formatting inconsistency in app_dial.c and make a warning message more useful about what frame type could not be written. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-04Replace ast_log(LOG_DEBUG, ...) with ast_debug()Paul Belanger
(closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03Asterisk media architecture conversion - no more format bitfieldsDavid Vossel
This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03Merged revisions 305923 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305923 | rmudgett | 2011-02-02 18:24:40 -0600 (Wed, 02 Feb 2011) | 24 lines Merged revisions 305889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines Merged revisions 305888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null terminator in the buffer length. When the frame is queued it is copied. If the null terminator is not part of the frame buffer length, the receiver could see garbage appended onto it. * Add channel lock protection with ast_sendtext(). * Fixed AMI SendText action ast_sendtext() return value check. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-02Replacing doc/* and asterisk.pdf with wiki linksAndrew Latham
Adding links to http(s)://wiki.asterisk.org git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-01Add's two features to confbridge: confbridge kick, and confbridge list.Brett Bryant
(closes issue #14389) (closes issue #18007) Reported by: jcollie Patches: 0001-Fix-up-bridging-module-so-that-menuselect-works.patch uploaded by jcollie (license 412) 0002-Add-confbridge-list-and-confbridge-kick-CLI-comm.patch uploaded by jcollie (license 412) Tested by: file Review: https://reviewboard.asterisk.org/r/1084/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-31Merged revisions 305254 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305254 | qwell | 2011-01-31 17:07:00 -0600 (Mon, 31 Jan 2011) | 24 lines Merged revisions 305253 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305253 | qwell | 2011-01-31 16:59:34 -0600 (Mon, 31 Jan 2011) | 17 lines Merged revisions 305252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) | 10 lines Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers already had code to prevent this. The attempt that app_dial was making to prevent it was not correct, so I fixed that. (closes issue #18371) Reported by: gbour Patches: 18371.patch uploaded by gbour (license 1162) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-31Merged revisions 304985 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304985 | tilghman | 2011-01-31 01:27:13 -0600 (Mon, 31 Jan 2011) | 16 lines Merged revisions 304978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304978 | tilghman | 2011-01-31 01:25:14 -0600 (Mon, 31 Jan 2011) | 9 lines Merged revisions 304952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304952 | tilghman | 2011-01-31 00:54:45 -0600 (Mon, 31 Jan 2011) | 2 lines Fix compilation when ODBC_STORAGE is defined. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-30Add Function and Application Relationships to documentationAndrew Latham
Add and extend the see-also sections to the documentation for applications and functions in an effort to expand the online documentation of the wiki. Also check for and update any links to moved documentation in the doc folder. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-29Merged revisions 304777 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304777 | seanbright | 2011-01-29 13:09:37 -0500 (Sat, 29 Jan 2011) | 22 lines Merged revisions 304776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304776 | seanbright | 2011-01-29 13:08:14 -0500 (Sat, 29 Jan 2011) | 15 lines If we fail to allocate our announcement objects, make sure we don't leak objects. The majority of this patch was committed already in r304726 and r304729. (issue #18225) Reported by: kenji (issue #18444) Reported by: junky (closes issue #18343) Reported by: kobaz Patches: meetme-refs.diff uploaded by kobaz (license 834) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304778 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-29Merged revisions 304774 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304774 | seanbright | 2011-01-29 12:54:43 -0500 (Sat, 29 Jan 2011) | 16 lines Merged revisions 304773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304773 | seanbright | 2011-01-29 12:51:28 -0500 (Sat, 29 Jan 2011) | 9 lines When we pass the S() or L() options to MeetMe, make sure that we honor C as well. Without this patch, if the user was kicked from the conference via the S() or L() mechanism, we would just hang up on them even if we also passed C (continue in dialplan when kicked). With this patch we honor the C flag in those cases. (closes issue #17317) Reported by: var ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-29Merged revisions 304730 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304730 | seanbright | 2011-01-29 12:15:27 -0500 (Sat, 29 Jan 2011) | 22 lines Merged revisions 304729 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304729 | seanbright | 2011-01-29 12:01:51 -0500 (Sat, 29 Jan 2011) | 15 lines Make sure that we unref the correct object when ejecting the most recent caller. Currently, when we kick the last user to enter, we decrement our own reference count which results in a crash when we kick another user or when we exit the conference ourselves. This will fix #18225 in 1.8 and trunk, but that particular bug does not exist in 1.6.2. (closes issue #18225) Reported by: kenji Patches: issue18225.patch uploaded by seanbright (license 71) Tested by: seanbright ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-29Merged revisions 304727 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304727 | seanbright | 2011-01-29 11:28:27 -0500 (Sat, 29 Jan 2011) | 16 lines Merged revisions 304726 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304726 | seanbright | 2011-01-29 11:26:57 -0500 (Sat, 29 Jan 2011) | 9 lines Fix user reference leak in MeetMe. We were unlinking the user from the conferences user container, but not decrementing the reference count of the user as well, resulting in a leak. (closes issue #18444) Reported by: junky Tested by: seanbright ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304728 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-28Merged revisions 304683 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304683 | seanbright | 2011-01-28 17:54:23 -0500 (Fri, 28 Jan 2011) | 16 lines Merged revisions 304659,304682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304659 | seanbright | 2011-01-28 16:22:09 -0500 (Fri, 28 Jan 2011) | 5 lines Don't leak references if we can't create a pseudo channel for mixing in MeetMe. If there was a problem allocating a pseudo channel when building our meetme, we weren't destroying our user container or destroying the mutexes that we created. ........ r304682 | seanbright | 2011-01-28 17:38:05 -0500 (Fri, 28 Jan 2011) | 2 lines Revert part of the previous commit that snuck in. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-26Add option to followme to delay answer until ready to bridge call.Jeff Peeler
Followme answers an incoming call if it hasn't already been answered and starts MOH. Some poorly designed autodialers see the answer and start playing their message to the hold music. The 'N' option has been added to indicate ringing and not answer until the call is accepted. (closes issue #18479) Reported by: ianc Patches: trunk_followme.diff uploaded by ianc (license 998) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25Merged revisions 303678 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303678 | jpeeler | 2011-01-25 11:02:38 -0600 (Tue, 25 Jan 2011) | 33 lines Merged revisions 303677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303677 | jpeeler | 2011-01-25 10:59:28 -0600 (Tue, 25 Jan 2011) | 26 lines Merged revisions 303676 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011) | 20 lines Fix voicemail sequencing for file based storage. A previous change was made to account for when the number of voicemail messages exceeds the max limit to be handled properly, but it caused gaps in the messages to not be properly handled. This has now been resolved. In later non 1.4 branches, it appears that resequencing wasn't even occurring due from what appears and accidental code removal. (closes issue #18498) Reported by: JJCinAZ Patches: bug18498v2.patch uploaded by jpeeler (license 325) (closes issue #18486) Reported by: bluefox Patches: bug18486.patch uploaded by jpeeler (license 325) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-24Merged revisions 303549 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303549 | russell | 2011-01-24 14:51:37 -0600 (Mon, 24 Jan 2011) | 45 lines Merged revisions 303548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines Merged revisions 303546 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines Fix channel redirect out of MeetMe() and other issues with channel softhangup. Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped working properly. This issue includes a patch that resolves the issue by removing a call to ast_check_hangup() from app_meetme.c. I left that in my patch, as it doesn't need to be there. However, the rest of the patch fixes this problem with or without the change to app_meetme. The key difference between what happens before and after this patch is the effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(), ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme sees this which causes it to exit as intended. Checking ast_check_hangup() caused app_meetme to exit earlier in the process, and the target of the redirect saw the condition where ast_read() returned NULL. Removing ast_check_hangup() works around the issue in app_meetme, but doesn't solve the issue if another application did the same thing. There are also other edge cases where if an application finishes at the same time that a redirect happens, the target of the redirect will think that the channel hung up. So, I made some changes in pbx.c to resolve it at a deeper level. There are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to abort the hangup process. My patch extends this to remove the END_OF_Q frame from the channel's read queue, making the "abort hangup" more complete. This same technique was used in every place where a softhangup flag was cleared. (closes issue #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: https://reviewboard.asterisk.org/r/1082/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-20Merged revisions 303009 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303009 | jpeeler | 2011-01-20 11:10:32 -0600 (Thu, 20 Jan 2011) | 21 lines Merged revisions 303008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303008 | jpeeler | 2011-01-20 11:07:44 -0600 (Thu, 20 Jan 2011) | 14 lines Merged revisions 303007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011) | 8 lines Add new queue strategy to preserve behavior for when queue members moved to ao2. Add queue strategy called "rrordered" to mimic old behavior from when queue members were stored in a linked list. ABE-2707 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-20Merged revisions 302921 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302921 | russell | 2011-01-20 10:12:15 -0600 (Thu, 20 Jan 2011) | 9 lines Merged revisions 302920 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302920 | russell | 2011-01-20 10:11:58 -0600 (Thu, 20 Jan 2011) | 2 lines Resolve a compiler warning. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302922 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-20Merged revisions 302918 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302918 | lmadsen | 2011-01-20 09:45:39 -0600 (Thu, 20 Jan 2011) | 16 lines Merged revisions 302917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302917 | lmadsen | 2011-01-20 09:42:05 -0600 (Thu, 20 Jan 2011) | 8 lines Option L() is milliseconds, not seconds. > Change the verbose output of option L() to say milliseconds and not seconds > as the value is in milliseconds. > > (closes issue #18264) > Reported by: jacco > Patches: > app_dial_patch.txt uploaded by lmadsen (license 10) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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