asterisk

Commit Graph

Author	SHA1	Message	Date
Leif Madsen	380e0e3e2d	Merged revisions 319367 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319367 \| lmadsen \| 2011-05-17 07:53:50 -0500 (Tue, 17 May 2011) \| 10 lines Don't create [general] voicemail context when using users.conf Prior to this patch, app_voicemail would create a [general] context when parsing users.conf. (closes issue #18891) Reported by: pdugas Patches: app_voicemail-ignore-general.patch uploaded by pdugas (license 1222) app_voicemail-ignore-general-style-guidelines.patch uploaded by seanbright (license 71) Tested by: pdugas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	892b7a2efd	Merged revisions 318671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318671 \| alecdavis \| 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) \| 30 lines Fix directed group pickup feature code *8 with pickupsounds enabled Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues. 1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked. 2). dialplan applications for directed_pickups shouldn't beep. 3). feature code for directed pickup should beep on success/failure if configured. Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite. Moved app_directed:pickup_do() to features:ast_do_pickup(). Functions below, all now use the new ast_do_pickup() app_directed_pickup.c: pickup_by_channel() pickup_by_exten() pickup_by_mark() pickup_by_part() features.c: ast_pickup_call() (closes issue #18654) Reported by: Docent Patches: ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett Review: https://reviewboard.asterisk.org/r/1185/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	6df3b851e3	Merged revisions 317969 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317969 \| russell \| 2011-05-06 16:49:01 -0500 (Fri, 06 May 2011) \| 10 lines Use the right variable to print the time in a debug message. The original patch also increased some buffer sizes, but that was already done in this version. (closes issue #17034) Reported by: sysreq Patches: asterisk-issue-17034.patch uploaded by sysreq (license 1009) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317970 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	d05e5281da	Merged revisions 317967 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317967 \| russell \| 2011-05-06 16:38:54 -0500 (Fri, 06 May 2011) \| 2 lines Fix some more "set but unused" compiler warnings. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317968 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	4fc020c965	Add the Uniqueid header to Userevent. (closes issue #16962) Reported by: jlpedrosa Patches: patch.diff uploaded by jlpedrosa (license 1002) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317915 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	892953466b	Merged revisions 317584 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r317584 \| twilson \| 2011-05-06 01:18:53 -0700 (Fri, 06 May 2011) \| 20 lines Merged revisions 317575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r317575 \| twilson \| 2011-05-06 01:04:17 -0700 (Fri, 06 May 2011) \| 13 lines Merged revisions 317574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317574 \| twilson \| 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011) \| 6 lines Re-fix queue round-robin This part of the change for r315596 was incorrect. No bridge occurs when doing a roundrobin dial and no one answers, so this code shouldn't have been removed. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317596 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	0ea3d21929	Merged revisions 317427 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317427 \| russell \| 2011-05-05 16:58:45 -0500 (Thu, 05 May 2011) \| 7 lines Fix potential memory leak, and use of uninitialized memory. (closes issue #16476) Reported by: junky Patches: M16476.diff uploaded by junky (license 177) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	7a2103efa6	Merged revisions 317336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317336 \| russell \| 2011-05-05 14:55:58 -0500 (Thu, 05 May 2011) \| 7 lines Increase buffer size to be PATH_MAX for a path. (closes issue #19239) Reported by: byronclark Patches: queue_announce_length.patch uploaded by byronclark (license 1200) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 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
Sean Bright	ed1ee072b8	Merged revisions 304774 via svnmerge from 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	15 years ago
Sean Bright	e229e9f010	Merged revisions 304730 via svnmerge from 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	15 years ago
Sean Bright	07b49f3adf	Merged revisions 304727 via svnmerge from 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	15 years ago
Sean Bright	c5cf436a92	Merged revisions 304683 via svnmerge from 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	15 years ago
Jeff Peeler	1c60cead78	Add option to followme to delay answer until ready to bridge call. 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	15 years ago
Jeff Peeler	d3c7a68982	Merged revisions 303678 via svnmerge from 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	15 years ago
Russell Bryant	092134399c	Merged revisions 303549 via svnmerge from 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	15 years ago
Jeff Peeler	a4fec286f8	Merged revisions 303009 via svnmerge from 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	15 years ago
Russell Bryant	7e42378131	Merged revisions 302921 via svnmerge from 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	15 years ago
Leif Madsen	876d5dede7	Merged revisions 302918 via svnmerge from 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	15 years ago
Sean Bright	59b2fbb984	Merged revisions 302834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302834 \| seanbright \| 2011-01-19 18:49:00 -0500 (Wed, 19 Jan 2011) \| 14 lines Merged revisions 302833 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302833 \| seanbright \| 2011-01-19 18:47:22 -0500 (Wed, 19 Jan 2011) \| 7 lines Support greetingsfolder as documented in voicemail.conf.sample. (closes issue #17870) Reported by: edhorton Patches: __20100816-app_voicemail-greetingsfolder-support.txt uploaded by lmadsen (license 10) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	563d973c11	Merged revisions 301177 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301177 \| pabelanger \| 2011-01-08 17:00:12 -0500 (Sat, 08 Jan 2011) \| 14 lines Merged revisions 301176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301176 \| pabelanger \| 2011-01-08 16:58:24 -0500 (Sat, 08 Jan 2011) \| 7 lines Indicate log level argument for Log() is not optional (closes issue #18586) Reported by: kshumard Patches: app_verbose.c.patch uploaded by kshumard (license 92) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301178 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	74e0a87776	Merged revisions 301090 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301090 \| qwell \| 2011-01-07 14:53:02 -0600 (Fri, 07 Jan 2011) \| 15 lines Merged revisions 301089 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301089 \| qwell \| 2011-01-07 14:52:00 -0600 (Fri, 07 Jan 2011) \| 8 lines Initialize useropts/adminopts in case there is no column in the realtime DB. (closes issue #18182) Reported by: dimas Patches: v1-18182.patch uploaded by dimas (license 88) Tested by: dimas ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	ac11bca7c0	Merged revisions 301047 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301047 \| jpeeler \| 2011-01-07 13:58:30 -0600 (Fri, 07 Jan 2011) \| 15 lines Merged revisions 301046 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301046 \| jpeeler \| 2011-01-07 13:57:42 -0600 (Fri, 07 Jan 2011) \| 8 lines Fix regression causing forwarding voicemails to not work with file storage. 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: https://origsvn.digium.com/svn/asterisk/trunk@301048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	3eec341083	Merged revisions 300955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300955 \| jpeeler \| 2011-01-07 11:24:14 -0600 (Fri, 07 Jan 2011) \| 21 lines Merged revisions 300951 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r300951 \| jpeeler \| 2011-01-07 11:23:37 -0600 (Fri, 07 Jan 2011) \| 14 lines Merged revisions 300918 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@300959 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Ruggles	19d14fe577	initialize playing_silence in struct initialization playing_silence was not initialized with the struct was initialized, it was being set after the fact which caused problems if something that relied on playing_silence being set was called too quickly (closes issue #18430) Reported by: stevebrandli Patches: externalivr.patch uploaded by thedavidfactor (license 903) Tested by: thedavidfactor, stevebrandli git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	1d48790cc2	Merged revisions 299989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299989 \| tilghman \| 2010-12-29 16:02:59 -0600 (Wed, 29 Dec 2010) \| 4 lines Quote arguments, just in case there's a space in a pathname. (Diagnosed by pabelanger on #asterisk-dev, fixed by me.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	addc30d3f1	Merged revisions 299865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r299865 \| pabelanger \| 2010-12-28 13:53:37 -0500 (Tue, 28 Dec 2010) \| 9 lines Merged revisions 299864 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r299864 \| pabelanger \| 2010-12-28 13:51:13 -0500 (Tue, 28 Dec 2010) \| 2 lines Documentation typo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	6765970cd2	Merged revisions 298685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298685 \| jpeeler \| 2010-12-16 17:31:50 -0600 (Thu, 16 Dec 2010) \| 16 lines Merged revisions 298684 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298684 \| jpeeler \| 2010-12-16 17:30:59 -0600 (Thu, 16 Dec 2010) \| 9 lines Merged revisions 298683 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@298686 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	6c0b904d17	Merged revisions 298598 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298598 \| jpeeler \| 2010-12-16 14:51:44 -0600 (Thu, 16 Dec 2010) \| 21 lines Merged revisions 298597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298597 \| jpeeler \| 2010-12-16 14:49:33 -0600 (Thu, 16 Dec 2010) \| 14 lines Merged revisions 298596 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@298599 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	997816819b	Merged revisions 297733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297733 \| tilghman \| 2010-12-06 18:29:26 -0600 (Mon, 06 Dec 2010) \| 22 lines Merged revisions 297713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297713 \| tilghman \| 2010-12-06 18:21:50 -0600 (Mon, 06 Dec 2010) \| 15 lines Merged revisions 297689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297689 \| tilghman \| 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010) \| 8 lines Don't create a Local channel if the target extension does not exist. (closes issue #18126) Reported by: junky Patches: followme.diff uploaded by junky (license 177) (partially restructured by me to avoid a possible memory leak) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	2b056c97cd	Merged revisions 297245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297245 \| russell \| 2010-12-02 07:20:19 -0600 (Thu, 02 Dec 2010) \| 20 lines Merged revisions 297229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297229 \| russell \| 2010-12-02 07:16:47 -0600 (Thu, 02 Dec 2010) \| 13 lines Merged revisions 297228 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297228 \| russell \| 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010) \| 6 lines Add "DAHDI" to a couple of app_meetme error messages. This is in response to some questions on IRC. To the user, there was nothing that made it obvious that this error had anything to do with DAHDI not being loaded. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	48ac3ea237	Merged revisions 296870 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296870 \| jpeeler \| 2010-11-30 18:28:16 -0600 (Tue, 30 Nov 2010) \| 18 lines Merged revisions 296869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296869 \| jpeeler \| 2010-11-30 18:24:58 -0600 (Tue, 30 Nov 2010) \| 11 lines Merged revisions 296868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296868 \| jpeeler \| 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010) \| 4 lines Properly restore backup information file when hanging up during message prepending. ABE-2654 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296871 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	72dc402f1f	Merged revisions 296787 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r296787 \| tilghman \| 2010-11-30 13:12:48 -0600 (Tue, 30 Nov 2010) \| 2 lines DOC: Conference number can be omitted; if omitted, all users in a meetme are listed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296788 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	758a671219	Merged revisions 296467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296467 \| tilghman \| 2010-11-27 04:40:22 -0600 (Sat, 27 Nov 2010) \| 12 lines Merged revisions 296466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r296466 \| tilghman \| 2010-11-27 04:39:01 -0600 (Sat, 27 Nov 2010) \| 5 lines 18 characters is too short for most date/times (20 is the usual, but we add more in case of greater precision). (closes issue #18369) Reported by: tnakonz ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Parisio	935930d8a3	Meetme use voicemail greet for join/leave announce Added option v(mailbox@[context]) which tells MeetMe where to look for a users greet file. If one does not exist it clears the v option and defers to the functionality of i/I as/if set by the MeetMe() command. Review: https://reviewboard.asterisk.org/r/1009/ (closes issue #18297) Reported by: parisioa Patches: meetme_final_patch_v.diff uploaded by parisioa (license 1153) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	712ba23185	Merged revisions 296002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296002 \| russell \| 2010-11-24 11:13:08 -0600 (Wed, 24 Nov 2010) \| 52 lines Merged revisions 296001 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296001 \| russell \| 2010-11-24 11:03:16 -0600 (Wed, 24 Nov 2010) \| 45 lines Merged revisions 296000 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296000 \| russell \| 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) \| 38 lines Handle failures building translation paths more effectively. The problem scenario occurred on a heavily loaded system that was using the codec_dahdi module and exceeded the hardware transcoding capacity. The failure mode at that point was not good. The report came in to us as an Asterisk lock-up. The "core show locks" shows a ton of threads locked up (but no obvious deadlock). Upon deeper investigation, when the system is in this state, the CPU was maxed out. The CPU was being consumed by the Asterisk logger spewing messages on every audio frame for calls set up after transcoder capacity was reached. The purpose of this patch is to make Asterisk handle failures to create a translation path in a more graceful manner. If we can't translate, then the call just needs to be dropped, as it's not going to work. These are the changes: 1) In set_format() of channel.c (which is called by set_read_format() and set_write_format()), it was ignoring if ast_translator_build_path() failed and returned NULL. It now pays attention to that case and returns a result reflecting failure. With this change in place, the bridging code will immediately detect a failure and end the bridge instead of proceeding to try to bridge frames that can't be translated and making channel drivers freak out by sending them frames in a format they weren't expecting. 2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was ignored. It is now reflected in the return value of the function. This didn't turn out to have any affect on the bug, but seemed like a good change to leave in. 3) In app_dial(), when only sending a call to a single endpoint, it will attempt to do some bridging of its own of early audio. It uses make_compatible() when it's going to do this. However, it ignored failure from make compatible. So, even with the fix from #1, if there was early audio going through app_dial, there would still be a period of invalid frames passing through. After detecting failure here, Dial() exits. ABE-2658 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296034 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	7c7486ad19	Merged revisions 295866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295866 \| rmudgett \| 2010-11-22 13:36:10 -0600 (Mon, 22 Nov 2010) \| 60 lines Merged revisions 295843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295843 \| rmudgett \| 2010-11-22 13:28:23 -0600 (Mon, 22 Nov 2010) \| 53 lines Merged revisions 295790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295790 \| rmudgett \| 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) \| 46 lines The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. To recreate the problem: 1) Party A calls Party B 2) Invoke CLI "channel redirect" command to redirect channel call leg associated with A. 3) All associated channels are hung up. Note that if the CLI command were done on the channel call leg associated with B it works. This regression was a result of the fix for issue #16946 (https://reviewboard.asterisk.org/r/740/). The regression affects all features that use an async goto to execute the dialplan because of an external event: Channel redirect, AMI redirect, SIP REFER, and FAX detection. The struct ast_channel._softhangup code is a mess. The variable is used for several purposes that do not necessarily result in the call being hung up. I have added doxygen comments to describe how the various _softhangup bits are used. I have corrected all the places where the variable was tested in a non-bit oriented manner. The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so the soft hangup requests that do not normally result in a hangup do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171) Reported by: SantaFox (closes issue #18185) Reported by: kwemheuer (closes issue #18211) Reported by: zahir_koradia (closes issue #18230) Reported by: vmarrone (closes issue #18299) Reported by: mbrevda (closes issue #18322) Reported by: nerbos Review: https://reviewboard.asterisk.org/r/1013/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	b54348691a	Merged revisions 295670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r295670 \| bbryant \| 2010-11-19 16:40:21 -0500 (Fri, 19 Nov 2010) \| 8 lines Patch for deadlock from ordering issue between channel/queue locks in app_queue (set_queue_variables). (closes issue #18031) Reported by: rain Review: https://reviewboard.asterisk.org/r/1018/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	6751c4f293	Merged revisions 294911 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294911 \| jpeeler \| 2010-11-12 15:14:43 -0600 (Fri, 12 Nov 2010) \| 11 lines Merged revisions 294910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294910 \| jpeeler \| 2010-11-12 15:14:23 -0600 (Fri, 12 Nov 2010) \| 4 lines Return correct error code if lock path fails. The recent changes to open_mailbox actually caused it to be fixed, but let's be consistent. Reported by alecdavis in asterisk-dev. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294912 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	03ec54e028	Merged revisions 294905 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294905 \| jpeeler \| 2010-11-12 14:52:06 -0600 (Fri, 12 Nov 2010) \| 30 lines Merged revisions 294904 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294904 \| jpeeler \| 2010-11-12 14:51:15 -0600 (Fri, 12 Nov 2010) \| 23 lines Merged revisions 294903 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294903 \| jpeeler \| 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010) \| 16 lines Fix regression causing abort in voicemail after opening a mailbox with no mesgs. In order to be more safe, some error handling code was changed to respect more error conditions including the potential memory allocation failure for deleted and heard message tracking introduced in 293004. However, last_message_index returns -1 for zero messages (perhaps as expected) and was triggering the stricter error checking. Because last_message_index is only called directly in one place, just return 0 from open_mailbox (for file based storage) when no messages are detected unless a real error has occurred. (closes issue #18240) Reported by: leobrown Patches: bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585) Tested by: pabelanger ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294907 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	4d76ac7a75	Merged revisions 293119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293119 \| jpeeler \| 2010-10-26 13:49:08 -0500 (Tue, 26 Oct 2010) \| 43 lines Merged revisions 293118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293118 \| jpeeler \| 2010-10-26 13:33:24 -0500 (Tue, 26 Oct 2010) \| 36 lines Merged revisions 293004 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293004 \| jpeeler \| 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010) \| 29 lines Fix inprocess_container in voicemail to correctly restrict max messages. The comparison function logic was off, so the number of sessions for a given mailbox were not being incremented properly. This problem caused the maximum number of messages per folder to not be respected when simultaneously leaving multiple voicemails just below the threshold. These problems should be fixed by the above, but just in case: Fixed resequence_mailbox to rely on the actual number of detected number of files in a directory rather than just assuming only 10 messages more than the maximum had been left. Also if more messages than the maximum are deleted they are actually removed now. The second purpose of this commit should have been separated out probably, but is related to the above. Again, if the number of messages in a given voicemail folder exceeds the maximum set limit make sure to allocate enough space for the deleted and heard index tracking array. A few random fixes: There was a forgotten decrement of the inprocess count in imap_store_file. When using IMAP storage, do not look in the directory where file based storage messages may still reside and influence the message count. Ensure to use only the first format in sendmail. ABE-2516 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	6abb7611f0	Merged revisions 292436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292436 \| pabelanger \| 2010-10-20 20:21:59 -0400 (Wed, 20 Oct 2010) \| 8 lines Application not properly unregister in voicemail (closes issue #18128) Reported by: junky Patches: vm_unregister.diff uploaded by junky (license 177) Tested by: pabelanger, lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292443 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	8da2aa88bb	Merged revisions 292413 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r292413 \| pabelanger \| 2010-10-20 20:07:17 -0400 (Wed, 20 Oct 2010) \| 24 lines Merged revisions 292412 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292412 \| pabelanger \| 2010-10-20 20:05:45 -0400 (Wed, 20 Oct 2010) \| 17 lines Merged revisions 292411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292411 \| pabelanger \| 2010-10-20 20:00:51 -0400 (Wed, 20 Oct 2010) \| 10 lines Record priv-recordintro as sln, not gsm This removes the gsm->sln step when transcoding priv-recordintro. (closes issue #18176) Reported by: pabelanger Patches: chan_sip.diff uploaded by pabelanger (license 224) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292414 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	96519117bb	Merged revisions 292227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r292227 \| jpeeler \| 2010-10-18 16:55:46 -0500 (Mon, 18 Oct 2010) \| 25 lines Merged revisions 292226 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292226 \| jpeeler \| 2010-10-18 16:54:38 -0500 (Mon, 18 Oct 2010) \| 18 lines Merged revisions 292223 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292223 \| jpeeler \| 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010) \| 11 lines Fix improper operator key acceptance and clean up temp recording files. This is a fix for when pressing the operator key after recording an unavailable, busy, name, or temporary message in mailbox options. The operator key should not be accepted here, but should be allowed during the message recording. If the operator key is pressed during ensure the file is saved or deleted as apporopriate. Also, ensure removal of temporary recorded files after an early hang up or when message acceptance confirmation times out. ABE-2518 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292228 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Stefan Schmidt	444d30b434	Report what extension called a failed macro Add the extension and context of the calling channel to the log output if a macro could not be found. (closes issue #18112) Reported by: prado Patches: app_macro-info.diff uploaded by prado (license 510) Tested by: schmidts git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291361 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	0e8c87d9b0	Merged revisions 290614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r290614 \| rmudgett \| 2010-10-06 13:50:37 -0500 (Wed, 06 Oct 2010) \| 12 lines Merged revision 290613 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r290613 \| rmudgett \| 2010-10-06 13:42:41 -0500 (Wed, 06 Oct 2010) \| 5 lines Eliminate a redundant test for AST_CONTROL_REDIRECTING. Eliminate redundant test for AST_CONTROL_REDIRECTING that prevents running the redirecting interception macro if it is defined. .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@290615 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	de22aaa413	Merged revisions 290376 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r290376 \| dvossel \| 2010-10-05 14:56:29 -0500 (Tue, 05 Oct 2010) \| 16 lines Merged revisions 290375 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r290375 \| dvossel \| 2010-10-05 14:54:50 -0500 (Tue, 05 Oct 2010) \| 10 lines Fixes PickupChan() not working with full channel name. (closes issue #18011) Reported by: schern Patches: app_directed_pickup.c.2.patch uploaded by schern (license 995) app_directed_pickup.c.trunk.patch uploaded by schern (license 995) Tested by: schern, dvossel ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@290377 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	f1244fd3f8	Merged revisions 289875 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289875 \| tilghman \| 2010-10-01 23:46:43 -0500 (Fri, 01 Oct 2010) \| 22 lines Merged revisions 289874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289874 \| tilghman \| 2010-10-01 23:45:49 -0500 (Fri, 01 Oct 2010) \| 15 lines Merged revisions 289873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289873 \| tilghman \| 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010) \| 8 lines When forwarding a message, a prepend means that the filesystem will always have a better copy. (closes issue #17803) Reported by: dpetersen Patches: 20100923__issue17803.diff.txt uploaded by tilghman (license 14) Tested by: dpetersen ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289876 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	f609c4f13a	Merged revisions 289426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289426 \| russell \| 2010-09-30 10:39:45 -0500 (Thu, 30 Sep 2010) \| 22 lines Merged revisions 289425 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289425 \| russell \| 2010-09-30 10:37:29 -0500 (Thu, 30 Sep 2010) \| 15 lines Merged revisions 289424 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289424 \| russell \| 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010) \| 8 lines Fix a crash in app_sms. Since the data being passed to the generator callback is on the stack of the SMS() application, we must ensure that the generator is stopped before the application exits. ABE-2587 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	7157b48150	Merged revisions 289104 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289104 \| tilghman \| 2010-09-28 13:18:43 -0500 (Tue, 28 Sep 2010) \| 4 lines Solaris compatibility fixes Review: https://reviewboard.asterisk.org/r/942/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	851141c131	Merged revisions 288079-288080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288079 \| rmudgett \| 2010-09-21 15:29:51 -0500 (Tue, 21 Sep 2010) \| 2 lines Protect channel access in CONNECTED_LINE and REDIRECTING interception macro launch code. ........ r288080 \| rmudgett \| 2010-09-21 15:29:59 -0500 (Tue, 21 Sep 2010) \| 8 lines Simplify locking code for REDIRECTING interception macro when forwarding a call. Simplified the locking code by using a local copy of the redirecting party information in app_dial.c:do_forward() and app_queue.c:wait_for_answer() for launching the REDIRECTING interception macro when a call is forwarded. Reduced the lock time of the 'o->chan' and 'in' channels. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288081 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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3957 Commits (bac5a51e2121a63549941f3a9ce7f7a4c50454ca)