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2011-06-06Remove Unused Var Warning rt_handle_member_recordirroot1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322128 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-06Refactor rt_handle_member_recordirroot1-14/+11
Review: https://reviewboard.asterisk.org/r/1172 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322111 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-01Merged revisions 321537 via svnmerge from bbryant1-6/+11
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321537 | bbryant | 2011-06-01 16:10:02 -0400 (Wed, 01 Jun 2011) | 8 lines This patch fixes an issue with using the wrong voicemail folders with greetings. (closes issue #17871) Reported by: edhorton Patches: digium_bug_17871_2 uploaded by fhackenberger (license 592) Tested by: edhorton, fhackenberger ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321538 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-27Merged revisions 321337 via svnmerge from rmudgett1-3/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 Also revert -r321331 and -r321332. ........ r321337 | rmudgett | 2011-05-27 17:06:43 -0500 (Fri, 27 May 2011) | 7 lines The app_privacy args have undocumented "options" position, interferes with "context" position. * Add documention for unused "options" position to match existing code. (closes issue #19273) Reported by: mdavenport ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321338 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-27Merged revisions 321330 via svnmerge from rmudgett1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321330 | rmudgett | 2011-05-27 16:31:25 -0500 (Fri, 27 May 2011) | 8 lines The app_privacy args have undocumented "options" position, interferes with "context" position. * Add documention for unused "options" position to match existing code. The trunk(v1.10) version will remove the unused options position. (closes issue #19273) Reported by: mdavenport ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321331 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-25Merged revisions 320823 via svnmerge from rmudgett4-3/+43
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines The AMI Newstate event contains different information between v1.4 and v1.8. The addition of connected line support in v1.8 changes the behavior of the channel caller ID somewhat. The channel caller ID value no longer time shares with the connected line ID on outgoing call legs. The timing of some AMI events/responses output the connected line ID as caller ID. These party ID's are now separate. * The ConnectedLineNum and ConnectedLineName headers were added to many AMI events/responses if the CallerIDNum/CallerIDName headers were also present. (closes issue #18252) Reported by: gje Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1227/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@320825 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-20Merged revisions 320237 via svnmerge from rmudgett1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r320237 | rmudgett | 2011-05-20 15:49:03 -0500 (Fri, 20 May 2011) | 27 lines Merged revisions 320236 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r320236 | rmudgett | 2011-05-20 15:44:54 -0500 (Fri, 20 May 2011) | 20 lines Merged revisions 320235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011) | 13 lines The meetme CLI command completion leaves conferences mutex locked. When issuing a meetme kick CLI command and an invalid (non-existent) conference number is specified, pressing Tab leaves the conferences mutex locked and, therefore, all conferences deadlock. Add missing unlock. (closes issue #19336) Reported by: zvision Patches: app_meetme.diff uploaded by zvision (license 798) ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@320238 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-20Merged revisions 320162 via svnmerge from jrose1-1/+15
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320162 | jrose | 2011-05-20 13:12:21 -0500 (Fri, 20 May 2011) | 15 lines Fixes an imapfolder related crash imapfolders being set in the general section of voicemail would cause the inbox folder name to change. Since sound file names are made based on the names of the folders, this would cause the audio related to that folder name to change and if Asterisk attempted to play it, the channel would instantly hang up when the audio file couldn't be found. This patch searches for the name of the folder first to leave existing behavior in tact and if that fails, it uses the normal inbox name to get the sound file instead. (closes issue #16104) Reported by: blkline Review: https://reviewboard.asterisk.org/r/1215/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@320178 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-20Merged revisions 320007 via svnmerge from rmudgett1-23/+24
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320007 | rmudgett | 2011-05-20 11:19:01 -0500 (Fri, 20 May 2011) | 2 lines Change some variable names to make pickup code easier to understand. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@320013 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-20Merged revisions 319997 via svnmerge from rmudgett1-68/+60
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319997 | rmudgett | 2011-05-20 10:48:25 -0500 (Fri, 20 May 2011) | 25 lines Crash when using directed pickup applications. The directed pickup applications can cause a crash if the pickup was successful because the dialplan keeps executing. This patch does the following: * Completes the channel masquerade on a successful pickup before the application returns. The channel is now guaranteed a zombie and must not continue executing the dialplan. * Changes the return value of the directed pickup applications to return zero if the pickup failed and nonzero(-1) if the pickup succeeded. * Made some code optimizations that no longer require re-checking the pickup channel to see if it is still available to pickup. (closes issue #19310) Reported by: remiq Patches: issue19310_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, remiq, rmudgett Review: https://reviewboard.asterisk.org/r/1221/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319998 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-18Merged revisions 319529 via svnmerge from twilson1-0/+11
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319529 | twilson | 2011-05-18 13:05:34 -0700 (Wed, 18 May 2011) | 24 lines Merged revisions 319528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r319528 | twilson | 2011-05-18 13:02:06 -0700 (Wed, 18 May 2011) | 17 lines Merged revisions 319527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011) | 10 lines Fix app_dial ring groups Revert part of r315643. We need to remove the datastore here as well. The code in bridging code will catch anything that app_dial might miss. (closes issue #19311) Reported by: mspuhler Patches: issue_19311_no_answer.diff uploaded by elguero (license 37) ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319530 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-17Merged revisions 319367 via svnmerge from lmadsen1-0/+3
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: http://svn.digium.com/svn/asterisk/trunk@319368 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-12Merged revisions 318671 via svnmerge from alecdavis1-61/+64
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: http://svn.digium.com/svn/asterisk/trunk@318672 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-06Merged revisions 317969 via svnmerge from russell1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@317970 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-06Merged revisions 317967 via svnmerge from russell1-6/+0
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: http://svn.digium.com/svn/asterisk/trunk@317968 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-06Add the Uniqueid header to Userevent.russell1-1/+6
(closes issue #16962) Reported by: jlpedrosa Patches: patch.diff uploaded by jlpedrosa (license 1002) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317915 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-06Merged revisions 317584 via svnmerge from twilson1-0/+11
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: http://svn.digium.com/svn/asterisk/trunk@317596 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-05Merged revisions 317427 via svnmerge from russell1-2/+6
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: http://svn.digium.com/svn/asterisk/trunk@317428 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-05Merged revisions 317336 via svnmerge from russell1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@317337 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-04Merged revisions 316831 via svnmerge from rmudgett1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@316832 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-04Merged revisions 316709 via svnmerge from seanbright1-2/+4
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: http://svn.digium.com/svn/asterisk/trunk@316711 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-04Merged revisions 316650 via svnmerge from dvossel1-2/+3
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: http://svn.digium.com/svn/asterisk/trunk@316657 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-04Merged revisions 316476 via svnmerge from seanbright1-0/+5
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: http://svn.digium.com/svn/asterisk/trunk@316477 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-03Merged revisions 316331 via svnmerge from russell1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@316333 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-03Merged revisions 316265 via svnmerge from russell8-34/+18
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: http://svn.digium.com/svn/asterisk/trunk@316293 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-02Formatting change, remove red blobspabelanger1-7/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316054 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-27Makes the new ConfBridge join and leave sounds be used by default rather ↵dvossel1-2/+2
than beep and beeperr. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@315856 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-26Merged revisions 315644 via svnmerge from twilson2-19/+0
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: http://svn.digium.com/svn/asterisk/trunk@315670 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-26Merged revisions 315452 via svnmerge from rmudgett1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@315453 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-21New HD ConfBridge conferencing application.dvossel4-335/+3560
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: http://svn.digium.com/svn/asterisk/trunk@314598 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-19Merged revisions 314203 via svnmerge from lmadsen1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@314204 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-19Add explanation of strange flag setup in app_meetme (stolen from Mark's ↵oej1-2/+3
message to asterisk-dev) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314158 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-18Merged revisions 314068 via svnmerge from rmudgett1-5/+8
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: http://svn.digium.com/svn/asterisk/trunk@314079 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-13Merged revisions 313517 via svnmerge from rmudgett1-64/+91
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: http://svn.digium.com/svn/asterisk/trunk@313527 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-11Merged revisions 313368-313369 via svnmerge from rmudgett1-10/+16
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: http://svn.digium.com/svn/asterisk/trunk@313383 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-07app_voicemail: close_mailbox change LOG_WARNING to LOG_NOTICEalecdavis1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@313003 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-05Minor change to 'L' option for meetme to include some verb statements for ↵jrose1-0/+8
the option. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@312756 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-01Merged revisions 312211 via svnmerge from alecdavis1-20/+63
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: http://svn.digium.com/svn/asterisk/trunk@312212 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-01Merged revisions 312117 via svnmerge from alecdavis1-1/+9
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: http://svn.digium.com/svn/asterisk/trunk@312118 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-28Merged revisions 311751 via svnmerge from russell1-0/+6
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: http://svn.digium.com/svn/asterisk/trunk@311752 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-23Merged revisions 311615 via svnmerge from bbryant1-5/+6
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: http://svn.digium.com/svn/asterisk/trunk@311616 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-22Merged revisions 311497 via svnmerge from dvossel1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@311498 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-18Adds an option to FollowMe that isn't useful for the bug it was made to ↵jrose1-3/+10
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: http://svn.digium.com/svn/asterisk/trunk@311427 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-18Merged revisions 311295 via svnmerge from rmudgett1-81/+200
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: http://svn.digium.com/svn/asterisk/trunk@311296 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-17Merged revisions 311197 via svnmerge from jrose1-1/+5
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: http://svn.digium.com/svn/asterisk/trunk@311198 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-11Mix Monitor: Now with r and t options.jrose1-56/+190
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310373 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-10Merged revisions 310142 via svnmerge from tilghman1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@310143 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-07Merged revisions 309858 via svnmerge from jrose1-1/+3
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: http://svn.digium.com/svn/asterisk/trunk@309859 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-04Merged revisions 309403 via svnmerge from diruggles1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@309404 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-22Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd ↵dvossel3-5/+5
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: http://svn.digium.com/svn/asterisk/trunk@308582 f38db490-d61c-443f-a65b-d21fe96a405b
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