3909 Commits (b13cfc92ec5ebef1e07875e2de020c9235d70b2d)

Author	SHA1	Message	Date
Tilghman Lesher	9a3fd9a994	Removing type attributes, as a change to menuselect makes them no longer necessary. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	d104b4e701	Add the attribute "type" to each "<use>" for menuselect. ... This matters only when autoconf fails to detect that weak linking is supported. External optional dependencies will become optional in both cases, as they are removed at compile time when not detected. However, runtime-optional modules are made mandatory when weak linking is not found. This change affects only the external optional dependencies; previously, they were incorrectly required when weak linking support was not detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003) Tested by: iasgoscouk git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326411 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Jordan	40babd5582	Patched voicemail user option for emailbody / emailsubject ... Incorporated changes per ASTERISK-16795; updated unit tests to check for vmu->emailbody / vmu->emailsubject (closes issue ASTERISK-16795) Reported by: mdeneen Tested by: mjordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@325877 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	1fe4351176	Fixed some error exit cleanup in app_queue.c. ... * Fixed error exit cleanup in app_queue.c copy_rules() and reload_queue_rules(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@325614 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	91b7dd582e	Response to QueueRule manager command does not contain ActionID if it was specified. ... * Add ActionID support as documented for the QueueRule AMI action. * Remove documentation for ActionID with the Queues AMI action. The output does not follow normal AMI response output and there is no place to put an ActionID header. (closes issue AST-602) Reported by: Vlad Povorozniuc Patches: jira_ast_602_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Vlad Povorozniuc, rmudgett Review: https://reviewboard.asterisk.org/r/1295/ JIRA SWP-3575 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@325610 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	3b216f2dc9	don't do native/remote bridging if a framehook is active on the channel ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@325537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kinsey Moore	a9a8c0fa05	ConfBridge does not handle hangup properly ... When playing back a prompt to a channel, confbridge neglects to check for hangup events causing lockup condititions for hangups that occur before actually joining the conference. This change ensures that the user is removed from the conference in the event of a premature hangup. Review: https://reviewboard.asterisk.org/r/1277/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324305 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	211af7820d	Fix typo in documentation. ... Pointed out by Vlad Povorozniuc git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324176 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	aec1979e7f	Remove potential deadlock in call pickup race. ... Deadlock is possible in ast_do_pickup() when holding the target channel lock and trying to get the chan channel lock. Also, holding the target lock when calling ast_channel_masquerade() is not a good idea because that routine does deadlock avoidance. * Removed the need to hold the target lock after marking the target with a datastore and getting the connected line data off of the target channel. * Moved can_pickup() to ast_can_pickup() in features.c. Now all the call pickup methods use the same basic call pickup availability check. Review: https://reviewboard.asterisk.org/r/1234/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	e0b2c103f6	Ring all queue with more than 255 agents will cause crash. ... 1. Create a ring-all queue with 500 permanent agents. 2. Call it. 3. Asterisk will crash. The watchers array in app_queue.c has a hard limit of 255. Bounds checking is not done on this array. No sane person should put 255 people in a ring-all queue, but we should not crash anyway. * Added bounds checking to the watchers array. JIRA AST-464 JIRA SWP-2903 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	ce51fcfb6b	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: https://origsvn.digium.com/svn/asterisk/branches/1.8@321537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	8323ef12bf	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: https://origsvn.digium.com/svn/asterisk/branches/1.8@321337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b1c397b5b7	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: https://origsvn.digium.com/svn/asterisk/branches/1.8@321330 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	59a41188a8	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: https://origsvn.digium.com/svn/asterisk/branches/1.8@320823 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	bf153ba6a4	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: https://origsvn.digium.com/svn/asterisk/branches/1.8@320237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	164f61d029	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: https://origsvn.digium.com/svn/asterisk/branches/1.8@320162 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	bf91f06f9f	Change some variable names to make pickup code easier to understand. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@320007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	7e3bf4936e	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: https://origsvn.digium.com/svn/asterisk/branches/1.8@319997 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	249f4b9022	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: https://origsvn.digium.com/svn/asterisk/branches/1.8@319529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	c23377d8f2	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/branches/1.8@319367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	87d80af96c	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/branches/1.8@318671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	10e40660c7	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/branches/1.8@317969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	91c8e4d297	Fix some more "set but unused" compiler warnings. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317967 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	6cf3280dd6	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/branches/1.8@317584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	bbf748d856	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/branches/1.8@317427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	a3d1ff1140	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/branches/1.8@317336 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	63579a892c	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/branches/1.8@316831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	6c3ea80a35	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/branches/1.8@316709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	eaf8673a16	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/branches/1.8@316650 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	aa43b12c24	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/branches/1.8@316476 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	1a8df4dc53	Resolve another warning. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	a82f1bb995	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/branches/1.8@316265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	734ca12381	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/branches/1.8@315644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	06223e643b	Add missing set of name valid flag when dialing. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@315452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	db02ef3704	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/branches/1.8@314203 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	bc620cd281	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/branches/1.8@314068 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	42882cd3bc	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/branches/1.8@313517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	dde33a1e01	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/branches/1.8@313369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	6dc376082d	Backport a restructuring change from trunk to make the next change stand out. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@313368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	8fe6967f1d	app_voicemail: close_mailbox change LOG_WARNING to LOG_NOTICE ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@313002 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	f6f5340777	Merged revisions 312762 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r312762 | jrose | 2011-04-05 09:11:36 -0500 (Tue, 05 Apr 2011) | 1 line Backporting trunk change to add verbosity to 'L' option in meetme ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@312765 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	62e679f784	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/branches/1.8@312211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	83aeb52dd0	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/branches/1.8@312117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	0a186e3f4f	Cross-reference VoiceMail() and VoiceMailMain() in the xml docs. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311751 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	51ce432d07	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/branches/1.8@311615 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	a00e99ec56	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/branches/1.8@311497 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	93601856b6	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/branches/1.8@311295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	ef01ba5ff2	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/branches/1.8@311197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	15641c348e	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/branches/1.8@310142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	4ad0ddf5e3	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/branches/1.8@309858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago