4015 Commits (e95b1dc220a9159b665feaa86023ec9b37f13112)

Author	SHA1	Message	Date
Richard Mudgett	5688637636	Fix crash during CDR update. ... The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to be called by different threads for the same channel. The channel driver thread and the PBX thread running dialplan. * Add lock protection around CDR API calls that access an ast_channel pointer. (closes issue ASTERISK-18836) Reported by: gpluser Review: https://reviewboard.asterisk.org/r/1628/ ........ Merged revisions 348362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	09fab9c14a	Fix ParkAndAnnounce to pass the CallerID to the announcing channel. ... ParkAndAnnounce tried to pass the CallerID to the announcing channel but the ID was wiped out by the channel masquerade done when parking the call. * Save the CallerID before parking the channel to pass it to the announcing channel. * Fixed a minor memory leak in ParkAndAnnounce. * Updated some ParkAndAnnounce log messages. ........ Merged revisions 348310 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	a3ac30d14c	Added support for all slin formats to app_originate ... Previously, app_originate could not originate a call into a non-8kHz conference bridge as the formats for non-8kHz slin codecs were not applied to the created channel. This patch adds all of the formats by default, such that if a created channel has a codec that supports a higher sampling rate, a translation path can be built between it and other channels. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	3d75189139	Fixed Asterisk crash when function QUEUE_MEMBER receives invalid input ... The function QUEUE_MEMBER has two required parameters (queuename, option). It was only checking for the presence of queuename. The patch checks for the existence of the option parameter and provides better error logging when invalid values are provided for the option parameter as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f71c13ed3d	Fix FollowMe CallerID on outgoing calls. ... The addition of the Connected Line support changed how CallerID is passed to outgoing calls. The FollowMe application was not updated to pass CallerID to the outgoing calls. * Fix FollowMe CallerID on outgoing calls. * Restructured findmeexec() to fix several memory leaks and eliminate some duplicated code. * Made check the return value of create_followme_number(). Putting a NULL into the numbers list is bad if create_followme_number() fails. * Fixed a couple uses of ast_strdupa() inside loops. * The changes to bridge_builtin_features.c fix a similar CallerID issue with the bridging API attended and blind transfers. (Not used at this time.) (closes issue ASTERISK-17557) Reported by: hamlet505a Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1612/ ........ Merged revisions 348101 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348102 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1ebeb108c8	Fix regressed behavior of queue set penalty to work without specifying 'in <queuename>' ... r325483 caused a regression in Asterisk 10+ that would make Asterisk segfault when attempting to set penalty on an interface without specifying a queue in the queue set penalty CLI command. In addition, no attempt would be made whatsoever to perform the penalty setting on all the queues in the core list with either the cli command or the non-segfaulting ami equivalent. This patch fixes that and also makes an attempt to document and rename some functions required by this command to better represent what they actually do. Oh yeah, and the use of this command without specifying a specific queue actually works now. Review: https://reviewboard.asterisk.org/r/1609/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ec1a9f4518	Fix: Meetme recording variables from realtime DB use null entries over channel variables ... Meetme would attempt to substitute the realtime values of RECORDING_FILE and RECORDING_FORMAT from the meetme db entry instead of using the channel variable set for those variables in spite of those database entries being NULL or even lacking a column to represent them. (closes issue ASTERISK-18873) Reported by: Byron Clark Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license 6157) ........ Merged revisions 347369 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	585499074d	Add regression tests for issue ASTERISK-18838. ... Review: https://reviewboard.asterisk.org/r/1572 Reviewed by: Matt Jordan ........ Merged revisions 347131 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	2ebbe7c31d	Move setting of voicemail zonetag and locale up a bit. ... The voicemail [general] zonetag and locale variables weren't loaded until after the mailboxes were initialized. This caused the settings to be unset for those mailboxes until a reload was performed. (closes issue ASTERISK-18838) Review: https://reviewboard.asterisk.org/r/1570 Reviewed by: Matt Jordan ........ Merged revisions 347111 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@347124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	c4a10a20be	Add missing sound_only_one config variable ... (closes issue ASTERISK-18895) Reported by: zvision Patches: conf_config_parser.diff (license #5755) patch uploaded by zvision git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@345882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	63e3243aab	Guarantee messages go into the right folders with multiple recipients ... Before, using the U flag in Voicemail with multiple recipients would put urgent messages in the INBOX folder for all users past the first thanks to a bug with the message copying function. This would also cause messages to fail to be sent if the INBOX directory hadn't been created for that mailbox yet. (closes issue ASTERISK-18245) Reported by: Matt Jordan (closes issue ASTERISK-18246) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1589/ ........ Merged revisions 345487 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@345488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	e6be04538c	Make queue log indicate if ADDMEMBER is paused for AMI and realtime. ... * Add parameter to queue log ADDMEMBER to indicate if the member is paused. (closes issue ASTERISK-18645) Reported by: garlew Patches: paused.diff (License #5337) patch uploaded by garlew Tested by: rmudgett, garlew Review: https://reviewboard.asterisk.org/r/1469/ ........ Merged revisions 345285 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@345290 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	e8fcda6a4d	Moves voicemail setup password entry to the end of the setup process. ... This change was made because forcegreeting and forcename settings in voicemail could be circumvented by hanging up after entering a password, because the only way voicemail currently observes whether a mailbox is new or not is by checking to see if the password is the same as the mailbox number or not. (closes issue ASTERISK-18282) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1581/ ........ Merged revisions 345062 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@345117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	572ab36b89	Fix app_macro.c MODULEINFO section termination. ... (closes issue ASTERISK-18848) Reported by: Tony Mountifield git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344557 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b2b0dfa615	Fix potential deadlock calling ast_call() with channel locks held. ... Fixed app_queue.c:ring_entry() calling ast_call() with the channel locks held. Chan_local attempts to do deadlock avoidance in its ast_call() callback and could deadlock if a channel lock is already held. ........ Merged revisions 344539 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344540 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	6692a0ad9e	Make AMI event AgentCalled get CallerID/ConnectedLine info from the incoming channel. ... It was strange that the AgentCalled AMI event would get most of its information from the incoming channel but then get the CallerID information from the outgoing channel. Before connected line support was added, this information was always the same at this point. (closes issue ASTERISK-18152) Reported by: Thomas Farnham Tested by: rmudgett ........ Merged revisions 344536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kinsey Moore	bf43b7c30e	Fix another incorrect case with meetme's PIN logic and add documentation ... This fixes an issue where a user of a dynamic conference was asked for a PIN twice. This also adds documentation to assist in future modifications to the piece of code responsible for PIN checking. (closes issue AST-670) ........ Merged revisions 344439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kinsey Moore	42686b23da	Fix pin parameter behavior regression in MeetMe ... The last time this code was touched (by me), a subtlety was missed based on the difference between needing to check a pin's validity and the need to prompt for a pin. (closes issue ASTERISK-18488) ........ Merged revisions 344102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@344103 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	3769a0e929	Add note about how Authenticate() application with option 'd' works. ... (closes issue ASTERISK-17422) Reported by: Leif Madsen ........ Merged revisions 343102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@343103 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kevin P. Fleming	48818ce81b	Modify comments in MeetMe application documentation about DAHDI. ... The MeetMe application documentation has some comments about usage of DAHDI, and they were a bit outdated relative to modern DAHDI releases. This patch changes the comment to just tell the user that a functional DAHDI timing source is required, and no longer mention 'dahdi_dummy', since that module does not exist in current DAHDI releases. ........ Merged revisions 342990 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	ba2c590c18	Use int for storing ao2_container_count instad of size_t ... AST-676 ........ Merged revisions 342435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	7e46d3ee91	Simplify queue membercount code ... Despite an ominous sounding comment stating that membercount was for "logged in" members only and thus we couldn't use ao2_container_count(), I could not find a single place in the code where that seemed to be accurate. The only time we decremented membercount was when we were marking something dead or actually removing it. The only places we incremented it were either after ao2_link(), or trying to correct for having set it to 0 during a reload. In every case where we were correcting the value, it seemed that we were trying to make the count actually match what ao2_container_count() would return. The only place I could find where we made a determination about something being "logged in" or not, we didn't trust the membercount, but instead looked at devicestate, paused, etc. This patch removes membercount, replaces its use with ao2_container_count, and manually adds the results of ao2_container_count to a "membercount" field for ast_data queue query results. This patch also would fix AST-676, but as it is slightly riskier than the previously committed fix, the two commits have been made separately. Reivew: https://reviewboard.asterisk.org/r/1541/ ........ Merged revisions 342383 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	1bc124ae58	Properly update membercount for reloaded members ... Since q->membercount is set to 0 before reloading, it is important to increment it again for reloaded members as well as added. (closes issue AST-676) Review: https://reviewboard.asterisk.org/r/1541/ ........ Merged revisions 342380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	55a92b7499	queues container needs locking when using the OBJ_NOLOCK flag ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342017 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	05f7d0c095	Remove some ref leaks and a return without unlock. ... There some resource leaks introduced in asterisk 10 make sure that locks are not held on return and we release ref's held. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341972 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	b0120e5e43	Revert Janitor patch 341920 For now ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341922 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	ec07b6448c	Whitespace Fixups / Add Braces ... This janitorial patch is related to work on RB1538 ........ Merged revisions 341906 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	3aa9147fc2	Add option to check state when state is unknown ... r341486 reverts r325483 this is a rework of the patch. optimize to minimize load. add option check_state_unknown to control whether a member with unknown device state is checked there is a small % chance that calls will be sent to the member when they on a call. app_queue will see a device with unknown state as available and does not try verify the state without this option enabled. Review: https://reviewboard.asterisk.org/r/1535/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	69d2d46275	Fix a performance regression introduced in r325483. ... The regression was caused by a call to ast_parse_device_state() in app_queue's ring_entry() function. The ast_parse_device_state() function eventually calls ast_channel_get_full() with a channel name prefix which causes it to walk the channel list causing massive lock contention and slow downs. This patch fixes the regression by removing the call to ast_parase_device_state() which should be unnecessary. Queue member device state should be maintained by device state events. Some users have seen instances where busy agents were called when they shouldn't have, which is the reason the call to ast_parse_device_state() was added. That change appears to have resolved that issue but also causes this performance regression. There may still be issues with queue member status, and if so, alternative methods should be investigated to resolve them. AST-695 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	824c5dbfa2	Multiple revisions 341108,341112 ... ........ r341108 | pabelanger | 2011-10-17 12:22:19 -0400 (Mon, 17 Oct 2011) | 2 lines Voicemail compiler flags are 'core' support ........ r341112 | pabelanger | 2011-10-17 12:23:33 -0400 (Mon, 17 Oct 2011) | 2 lines Fix previous commit ........ Merged revisions 341108,341112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341122 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	d3831204a9	Fixes some support level info so that it can be read by menuselect. ... (issue ASTERISK-18268) Review: https://reviewboard.asterisk.org/r/1525/ ........ Merged revisions 340863 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ec1778c05f	Update MeetMe p and X option documentation when interacting with the s option. ... ASTERISK-12175 changed the p and X options to not interfere with the s option when they are used together. It makes more sense for the s option to have priority for the DTMF '*' key since it cannot change its activation code. Otherwise, you could not use option s with the p or X options. JIRA AST-671 ........ Merged revisions 340470 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	63d4530e93	Merged revisions 340108 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct 2011) | 11 lines Load the proper XML documentation when multiple modules document the same application. This patch adds an optional "module" attribute to the XML documentation spec that allows the documentation processor to match apps with identical names from different modules to their documentation. This patch also fixes a number of bugs with the documentation processor and should make it a little more efficient. Support for multiple languages has also been properly implemented. ASTERISK-18130 Review: https://reviewboard.asterisk.org/r/1485/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340109 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	533a924ae5	Merged revisions 339776 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339776 | rmudgett | 2011-10-07 14:34:55 -0500 (Fri, 07 Oct 2011) | 5 lines Initialize option flags for SendURL application. (closes issue ASTERISK-18574) Reported by: marcelloceschia ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	7a2320945b	Merged revisions 339511 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339511 | rmudgett | 2011-10-05 12:01:01 -0500 (Wed, 05 Oct 2011) | 1 line Fix Dial F option notes formatting. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339512 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	15f3a3bee1	Merged revisions 339144 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339144 | lmadsen | 2011-10-03 14:54:52 -0500 (Mon, 03 Oct 2011) | 6 lines Make documentation for Dial() options 'F' and 'F()' more clear. (Closes issue ASTERISK-18646) Reported by: Physis Heckman Tested by: Richard Mudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@339145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	80a5b370ff	Merged revisions 338084 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338084 | pabelanger | 2011-09-27 16:10:13 -0400 (Tue, 27 Sep 2011) | 2 lines Upgrade app_macro to core ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338085 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	0764556d4f	Merged revisions 337973 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011) | 30 lines Fix deadlock when using dummy channels. Dummy channels created by ast_dummy_channel_alloc() should be destoyed by ast_channel_unref(). Using ast_channel_release() needlessly grabs the channel container lock and can cause a deadlock as a result. * Analyzed use of ast_dummy_channel_alloc() and made use ast_channel_unref() when done with the dummy channel. (Primary reason for the reported deadlock.) * Made app_dial.c:dial_exec_full() not call ast_call() holding any channel locks. Chan_local could not perform deadlock avoidance correctly. (Potential deadlock exposed by this issue. Secondary reason for the reported deadlock since the held lock was part of the deadlock chain.) * Fixed some uses of ast_dummy_channel_alloc() not checking the returned channel pointer for failure. * Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected by testing the bogus_chan value. * Fixed needlessly clearing a 1024 char auto array when setting the first char to zero is enough in manager.c:action_getvar(). (closes issue ASTERISK-18613) Reported by: Thomas Arimont Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Thomas Arimont ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337974 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	876cf3f78e	Merged revisions 337839 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337839 | irroot | 2011-09-23 10:34:03 +0200 (Fri, 23 Sep 2011) | 11 lines Make sure a CDR is on the stack for call in the Queue. Only let update_cdr act on the last CDR in the stack. In some circumstances [Attended transfer to queue] a CDR record is not inserted for this call where it should. (closes issue ASTERISK-18567) Review: https://reviewboard.asterisk.org/r/1266 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	cf4d5e585a	Revert previous commit ... New feature should be added into trunk, unfortunately it is too late for the Asterisk 10 branch. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	ee9db5269c	Revert commit r337261 ... This commit is for trunk not version 10 ----- Adds a timeout argument to app_originate the default is 30s this will be used if the timout supplied is invalid or no timeout is supplied. ----- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	94d511c5fa	More silly spacing changes ...	15 years ago
Tilghman Lesher	e71a6090f2	........ ...	15 years ago
Gregory Nietsky	edecb53861	Adds a timeout argument to app_originate ... the default is 30s this will be used if the timout supplied is invalid or no timeout is supplied. Contributed by: jacco (thank you for the work) Review: https://reviewboard.asterisk.org/r/1310/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Jordan	944cdaa94d	Merged revisions 337118 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) | 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	8285989c1c	Merged revisions 336716 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336716 | jrose | 2011-09-19 15:07:36 -0500 (Mon, 19 Sep 2011) | 7 lines Document applications that play audio and do not answer unanswered calls. This patch is part of an effort to document early media and its usage. If you are interested in contributing to this documentation effort, there are probably other applications worth documenting as well as an Asterisk wiki article at https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	6fd0d3805d	Merged revisions 336658 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011) | 31 lines Made Dial d and H options no longer immediately auto-answer the calling leg. The Dial d and H options break DTMF attended transfer atxferdropcall option. 1) Party A calls party B. 2) Party B does a DTMF attended transfer to Party C. If the dialplan uses the Dial d or H options to call Party C then the Dial application answers the call immediately before initiating the call leg to Party C. The premature answer causes the transfer code to not invoke the atxferdropcall=no behavior for a blonde transfer since Party C has "answered". The transfer code thinks that Party B has "consulted" with Party C when Party B hangs up and completes the transfer to Party A. Party A now hears ringback until Party C actually answers. ASTERISK-13294 Dial d option. ASTERISK-11067 Dial H option to disconnect before answer. The referenced issues made Dial answer with the d and H options because many SIP and ISDN phones cannot send DTMF before the call is connected. * Made require the dialplan to control when or if the call needs to be answered to use the Dial application d and H options. (The call is no longer surprise answered when using the Dial d or H options.) Review: https://reviewboard.asterisk.org/r/1381/ JIRA AST-623 JIRA AST-666 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336659 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	a687f85aa5	Merged revisions 336093 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336093 | irroot | 2011-09-15 17:46:21 +0200 (Thu, 15 Sep 2011) | 20 lines Locking order in app_queue.c causes deadlocks. a channel lock must never be held with the queues container lock held. the deadlock occured on masquerade. the queues container lock is a relic of the past the old queue module lock. with ao2 there is no need to hold this lock when dealing with members this patch removes unneeded locks. (closes issue ASTERISK-18101) (closes issue ASTERISK-18487) Reported by: Paul Rolfe, Jason Legault Tested by: irroot, Jason Legault, Paul Rolfe Reviewed by: Matthew Nicholson Review: https://reviewboard.asterisk.org/r/1402/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336094 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Olle Johansson	9795ec3ef0	Meetme: Introducing a new option "k" to kill a conference if there's only a single member left. ... When using Meetme as a modular call bridge from third party applications, it's handy to make it behave like a normal call bridge. When the second to last person exists, the last person will be kicked out of the conference when this option is enabled. (closes issue ASTERISK-18234) Review: https://reviewboard.asterisk.org/r/1376/ Patch by oej, sponsored by ClearIT, Solna, Sweden git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	d297a943e2	Merged revisions 335720 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335720 | rmudgett | 2011-09-13 17:10:15 -0500 (Tue, 13 Sep 2011) | 1 line Remove obsolete todo comment about PICKUPRESULT. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago