asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	a45d2f29c6	Merged revisions 316831 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316831 \| rmudgett \| 2011-05-04 13:51:40 -0500 (Wed, 04 May 2011) \| 9 lines Wait for leader with Music On Hold allows crosstalk between participants. Parenthesis in the wrong position. Regression from issue #14365 when expanding conference flags to use 64 bits. (closes issue #18418) Reported by: MrHanMan Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	51fc64d13a	Merged revisions 316709 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316709 \| seanbright \| 2011-05-04 12:15:32 -0400 (Wed, 04 May 2011) \| 22 lines Merged revisions 316708 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r316708 \| seanbright \| 2011-05-04 12:10:59 -0400 (Wed, 04 May 2011) \| 15 lines Merged revisions 316707 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r316707 \| seanbright \| 2011-05-04 12:08:50 -0400 (Wed, 04 May 2011) \| 8 lines If sox fails when processing a voicemail, don't delete the original file. (closes issue #18111) Reported by: sysreq Patches: issue18111_trunk.patch uploaded by seanbright (license 71) Tested by: seanbright ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316711 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	a3fd2b77b6	Merged revisions 316650 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316650 \| dvossel \| 2011-05-04 09:25:03 -0500 (Wed, 04 May 2011) \| 15 lines Merged revisions 316644 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r316644 \| dvossel \| 2011-05-04 09:23:39 -0500 (Wed, 04 May 2011) \| 9 lines Fixes one-way-audio when chanspy activated with the 'o' option (closes issue #18382) Reported by: jkister Patches: 0001-Bugfix-18382-one-way-audio-when-chanspy-activated.patch.txt uploaded by malin (license ) Tested by: firstsip, Greenlightcrm, malin, wdoekes, boroda, dvossel ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316657 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	c596329564	Merged revisions 316476 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316476 \| seanbright \| 2011-05-03 22:34:01 -0400 (Tue, 03 May 2011) \| 17 lines Merged revisions 316475 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r316475 \| seanbright \| 2011-05-03 22:23:01 -0400 (Tue, 03 May 2011) \| 10 lines Honor the C option to MeetMe when L is passed. This fixes a case that r304773 and friends missed. (closes issue #17317) Reported by: var Patches: meetme-continue-on-l_16218.diff uploaded by var (license 1227) Tested by: seanbright ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316477 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	277f9f46dc	Merged revisions 316331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316331 \| russell \| 2011-05-03 16:41:11 -0500 (Tue, 03 May 2011) \| 2 lines Resolve another warning. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316333 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	37aa52fd78	Merged revisions 316265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316265 \| russell \| 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) \| 5 lines Fix a bunch of compiler warnings generated by gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316293 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	7c3d14957b	Formatting change, remove red blobs git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316054 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	696c77c59e	Makes the new ConfBridge join and leave sounds be used by default rather than beep and beeperr. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	8d2a71877a	Merged revisions 315644 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315644 \| twilson \| 2011-04-26 14:39:01 -0700 (Tue, 26 Apr 2011) \| 32 lines Merged revisions 315643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315643 \| twilson \| 2011-04-26 14:27:44 -0700 (Tue, 26 Apr 2011) \| 25 lines Merged revisions 315596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315596 \| twilson \| 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) \| 18 lines Allow transfer loops without allowing forwarding loops We try to avoid the situation where two phones may be forwarded to each other causing an infinite loop by storing each dialed interface in a channel datastore and checking the list before dialing out. This works, but currently breaks situations like A calls B, A transfers B to C, B transfers C to A, and A transfers C to B. Since human interaction is happening here and not an automated forwarding loop, it should be allowed. This patch removes the dialed_interfaces datastore when a call is bridged (a suggestion from the brilliant mmichelson). If a call is being bridged, it should be safe to assume that we aren't stuck in a loop. Since we are now handling this is the bridge code, the previous attempts at handling it in app_dial and app_queue are removed. Review: https://reviewboard.asterisk.org/r/1195/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	abe0351e12	Merged revisions 315452 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315452 \| rmudgett \| 2011-04-26 13:00:34 -0500 (Tue, 26 Apr 2011) \| 1 line Add missing set of name valid flag when dialing. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	7f23115ad2	New HD ConfBridge conferencing application. 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	14 years ago
Leif Madsen	072970e1ab	Merged revisions 314203 via svnmerge from 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	14 years ago
Olle Johansson	0622568f15	Add explanation of strange flag setup in app_meetme (stolen from Mark's message to asterisk-dev) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	7c4fc0f0e8	Merged revisions 314068 via svnmerge from 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	14 years ago
Richard Mudgett	11852af23a	Merged revisions 313517 via svnmerge from 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	14 years ago
Richard Mudgett	663ed7fd5c	Merged revisions 313368-313369 via svnmerge from 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	14 years ago
Alec L Davis	1166d8dfa1	app_voicemail: close_mailbox change LOG_WARNING to LOG_NOTICE git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313003 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	5af547a619	Minor change to 'L' option for meetme to include some verb statements for the option. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312756 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	e59a051c3e	Merged revisions 312211 via svnmerge from 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	14 years ago
Alec L Davis	d07fb85bb8	Merged revisions 312117 via svnmerge from 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	14 years ago
Russell Bryant	c4c13423bf	Merged revisions 311751 via svnmerge from 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	14 years ago
Brett Bryant	c31d7b21ea	Merged revisions 311615 via svnmerge from 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	14 years ago
David Vossel	7902813301	Merged revisions 311497 via svnmerge from 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	14 years ago
Jonathan Rose	18a6c3a415	Adds an option to FollowMe that isn't useful for the bug it was made to 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	14 years ago
Richard Mudgett	4a8c77976c	Merged revisions 311295 via svnmerge from 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	14 years ago
Jonathan Rose	d956ecb96e	Merged revisions 311197 via svnmerge from 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	14 years ago
Jonathan Rose	6e36042f64	Mix Monitor: Now with r and t options. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310373 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	67c91388db	Merged revisions 310142 via svnmerge from 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	14 years ago
Jonathan Rose	3845fb50c0	Merged revisions 309858 via svnmerge from 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	14 years ago
David Ruggles	3cda82a379	Merged revisions 309403 via svnmerge from 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	14 years ago
David Vossel	d760e81f37	Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd 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	14 years ago
Jason Parker	551dac2eda	Merged revisions 308010 via svnmerge from 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	14 years ago
Richard Mudgett	b1db966684	Merged revisions 307962 via svnmerge from 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	14 years ago
Tilghman Lesher	7800a1c330	Merged revisions 307750 via svnmerge from 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	14 years ago
Jeff Peeler	8f7982f280	Add new manager action MeetmeListRooms. 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	14 years ago
Jeff Peeler	a46bfe67bd	Merged revisions 306967 via svnmerge from 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	14 years ago
Jeff Peeler	e2cdaf47bb	Merged revisions 306962 via svnmerge from 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	14 years ago
Jeff Peeler	9264ab00f5	Merged revisions 306866 via svnmerge from 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	14 years ago
Richard Mudgett	a8aeb04a9f	Add ISDN display ie text handling options to chan_dahdi.conf. 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	14 years ago
Jason Parker	0beeb00ef3	Merged revisions 306356 via svnmerge from 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	14 years ago
Richard Mudgett	4d8feab7fa	Merged revisions 306324 via svnmerge from 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	14 years ago
Paul Belanger	3556e4c2d4	Replace ast_log(LOG_DEBUG, ...) with ast_debug() (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	14 years ago
David Vossel	c26c190711	Asterisk media architecture conversion - no more format bitfields 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	14 years ago
Richard Mudgett	f71322f239	Merged revisions 305923 via svnmerge from 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	15 years ago
Andrew Latham	93bade5639	Replacing doc/* and asterisk.pdf with wiki links Adding links to http(s)://wiki.asterisk.org git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305843 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	eec87e3266	Add's two features to confbridge: confbridge kick, and confbridge list. (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	15 years ago
Jason Parker	6908539952	Merged revisions 305254 via svnmerge from 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	15 years ago
Tilghman Lesher	e3b475b0ad	Merged revisions 304985 via svnmerge from 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	15 years ago
Andrew Latham	f9c3b26241	Add Function and Application Relationships to documentation 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	15 years ago
Sean Bright	cc2c9442f6	Merged revisions 304777 via svnmerge from 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	15 years ago

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