asterisk

Commit Graph

Author	SHA1	Message	Date
Kinsey Moore	782cfdc775	Merged revisions 335346 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335346 \| kmoore \| 2011-09-12 09:22:15 -0500 (Mon, 12 Sep 2011) \| 17 lines Merged revisions 335341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335341 \| kmoore \| 2011-09-12 09:21:17 -0500 (Mon, 12 Sep 2011) \| 10 lines Ensure frames are not written to dialed channel if ringback is requested When a single channel was dialed and there was media to be forwarded to the calling channel, the media was written without regard for ringback causing silence to be heard in some circumstances. This regression was introduced when the meaning of "single" changed to mean only the number of channels dialed. (closes issue ASTERISK-18083) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	8b5ba33fe0	Merged revisions 335078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335078 \| mjordan \| 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) \| 29 lines Merged revisions 335064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 \| mjordan \| 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) \| 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	8017b65bb9	Merged revisions 335014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r335014 \| irroot \| 2011-09-09 09:23:53 +0200 (Fri, 09 Sep 2011) \| 9 lines Move code for VALID_EXTEN from app_readexten to func_dialplan Mark VALID_EXTEN deprecated. Review: https://reviewboard.asterisk.org/r/1396/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	5ad57732f5	Merged revisions 334621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334621 \| alecdavis \| 2011-09-07 20:14:50 +1200 (Wed, 07 Sep 2011) \| 9 lines Merged revisions 334620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334620 \| alecdavis \| 2011-09-07 20:12:49 +1200 (Wed, 07 Sep 2011) \| 2 lines peroid typo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	f090651138	Merged revisions 334455 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334455 \| irroot \| 2011-09-06 15:58:56 +0200 (Tue, 06 Sep 2011) \| 18 lines Merged revisions 334453 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334453 \| irroot \| 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) \| 13 lines Make SQL query in app_voicemail.c portable LIMIT is not portable. Regression from r312212 (closes issue ASTERISK-18255) Reported by: Leif Madsen Tested by: Leif Madsen Review: https://reviewboard.asterisk.org/r/1415/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	8a8baa1934	Revert r334472 due to properties going missing git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334515 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	4435439eda	Merged revisions 334455 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334455 \| irroot \| 2011-09-06 15:58:56 +0200 (Tue, 06 Sep 2011) \| 18 lines Merged revisions 334453 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334453 \| irroot \| 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) \| 13 lines Make SQL query in app_voicemail.c portable LIMIT is not portable. Regression from r312212 (closes issue ASTERISK-18255) Reported by: Leif Madsen Tested by: Leif Madsen Review: https://reviewboard.asterisk.org/r/1415/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	a91b1149b9	Merged revisions 333631 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333631 \| mjordan \| 2011-08-29 12:12:55 -0500 (Mon, 29 Aug 2011) \| 9 lines Merged revisions 333630 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333630 \| mjordan \| 2011-08-29 12:11:15 -0500 (Mon, 29 Aug 2011) \| 1 line Fixed improperly formatted TestEvent AMI message in app_voicemail ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333632 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	a721549656	Merged revisions 333370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333370 \| mjordan \| 2011-08-26 10:58:37 -0500 (Fri, 26 Aug 2011) \| 26 lines Merged revisions 333339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333339 \| mjordan \| 2011-08-26 08:36:36 -0500 (Fri, 26 Aug 2011) \| 20 lines Bug fixes for voicemail user emailsubject / emailbody. This code change fixes a few issues with the voicemail user override of emailbody and emailsubject, including escaping the strings, potential memory leaks, and not overriding the voicemail defaults. Revision 325877 fixed this for ASTERISK-16795, but did not fix it for ASTERISK-16781. A subsequent check-in prevented 325877 from being applied to 10. This check-in resolves both issues, and applies the changes to 1.8, 10, and trunk. (closes issue ASTERISK-16781) Reported by: Sebastien Couture Tested by: mjordan (closes issue ASTERISK-16795) Reported by: mdeneen Tested by: mjordan Review: https://reviewboard.asterisk.org/r/1374 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333371 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	436ceb827c	Merged revisions 333011 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333011 \| rmudgett \| 2011-08-23 13:15:49 -0500 (Tue, 23 Aug 2011) \| 19 lines Merged revisions 333010 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333010 \| rmudgett \| 2011-08-23 13:14:01 -0500 (Tue, 23 Aug 2011) \| 12 lines Memory Leak in app_queue The patch that was committed in the 1.6.x versions of Asterisk for ASTERISK-15862 actually fixed two issues. One was not applicable to 1.8 but the other is. queue_leak.patch fixes the portion applicable to 1.8. (closes issue ASTERISK-18265) Reported by: Fred Schroeder Patches: queue_leak.patch (license #5049) patch uploaded by mmichelson Tested by: Thomas Arimont ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333014 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b92dcb0c82	Merged revisions 332875,332878 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332875 \| rmudgett \| 2011-08-22 14:41:03 -0500 (Mon, 22 Aug 2011) \| 1 line Fix merge property. ................ r332878 \| rmudgett \| 2011-08-22 14:46:25 -0500 (Mon, 22 Aug 2011) \| 25 lines Merged revisions 332874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332874 \| rmudgett \| 2011-08-22 14:32:19 -0500 (Mon, 22 Aug 2011) \| 18 lines Reference leaks in app_queue. * Fixed load_realtime_queue() leaking a queue reference when it overwrites q when processing a realtime queue. (issue ASTERISK-18265) * Make join_queue() unreference the queue returned by load_realtime_queue() when it is done with the pointer. The load_realtime_queue() returns a reference to the just loaded realtime queue. * Fixed queues container reference leak in queues_data_provider_get(). * queue_unref() should not return q that was just unreferenced. * Made logic in __queues_show() and queues_data_provider_get() when calling load_realtime_queue() easier to understand. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	3b53a9cdb3	Merged revisions 332817 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332817 \| mjordan \| 2011-08-22 13:15:51 -0500 (Mon, 22 Aug 2011) \| 4 lines Review: https://reviewboard.asterisk.org/r/1364/ This update adds a new AMI event, TestEvent, which is enabled when the TEST_FRAMEWORK compiler flag is defined. It also adds initial usage of this event to app_voicemail. The TestEvent AMI event is used extensively by the voicemail tests in the Asterisk Test Suite. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	3e89d62884	Merged revisions 332654 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r332654 \| kmoore \| 2011-08-19 14:59:34 -0500 (Fri, 19 Aug 2011) \| 8 lines Make CONFBRIDGE_INFO behave more nicely CONFBRIDGE_INFO doesn't behave as well in edge cases as MEETME_INFO. With this patch, CONFBRIDGE_INFO should behave in a much more reasonable manner when presented with invalid conferences and keywords. Review: https://reviewboard.asterisk.org/r/1359/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	c9f65ece49	Merged revisions 331775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331775 \| mnicholson \| 2011-08-12 14:03:31 -0500 (Fri, 12 Aug 2011) \| 17 lines Merged revisions 331774 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331774 \| mnicholson \| 2011-08-12 14:01:27 -0500 (Fri, 12 Aug 2011) \| 11 lines Unlock the channel before calling update_queue. Holding the channel lock when calling update_queue which attempts to lock the queue lock can cause a deadlock. This deadlock involves the following chain: 1. hold chan lock -> wait queue lock 2. hold queue lock -> wait agent list lock 3. hold agent list lock -> wait chan list lock 4. hold chan list lock -> wait chan lock ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331776 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	39fe851e79	Merged revisions 331644 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331644 \| jrose \| 2011-08-12 11:18:57 -0500 (Fri, 12 Aug 2011) \| 9 lines Merged revisions 331635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331635 \| jrose \| 2011-08-12 10:49:17 -0500 (Fri, 12 Aug 2011) \| 1 line Fixes 32bit compilation warnings brought on by 331634 in app_dial and app_meetme ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	1a8069abe2	Merged revisions 331579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331579 \| qwell \| 2011-08-11 16:54:54 -0500 (Thu, 11 Aug 2011) \| 13 lines Merged revisions 331578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331578 \| qwell \| 2011-08-11 16:46:39 -0500 (Thu, 11 Aug 2011) \| 6 lines Use proper values for 64-bit option flags. Also, reusing bits es no bueno, so change the value of a duplicate. (issue ASTERISK-18239) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b99b1116be	Merged revisions 331265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331265 \| rmudgett \| 2011-08-09 18:12:49 -0500 (Tue, 09 Aug 2011) \| 22 lines Merged revisions 331248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331248 \| rmudgett \| 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011) \| 15 lines Misc minor items found in code. * Add some reentrancy protection in pbx.c when creating the contexts_table hash table. * Fix inverted test in chan_sip.c conditional code. * Fix uninitialized variable and use of the wrong variable in chan_iax2.c. * Fix test of return value in app_parkandannounce.c. Explicitly testing for -1 is bad if the function does not actually return that value when it fails. * Fixup some comments and add some curly braces in features.c. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	0f5ef2c781	Log queue member name when state_interface is set for ADDMEMBER and REMOVEMEMBER events app_queue logs the events ADDMEMBER and REMOVEMEMBER with the agent field set to the interface value rather than the membername value when a member is added with a state_interface value set. However all other member related queue events are logged with the membername when a state_interface is set. This patch makes these fields optionally more consistent and correct. (closes issue ASTERISK-14769) Review: https://reviewboard.asterisk.org/r/1286 Patch-by: Jamuel Starkey Tested-by: Kinsey Moore <kmoore@digium.com> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331037 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	d1a0938c99	app_queue: Add StateInterface to output of "queue show" and "QueueStatus" This patch adds the state_interface of the queue member struct to the output of "queue show" (CLI command) and "QueueStatus" (AMI action) when displaying relevant queue member information. For the AMI event message the variable StateInterface has been added. (closes issue ASTERISK-18071) Review: https://reviewboard.asterisk.org/r/1300/ Patch-by: Jamuel Starkey git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	b6a9795b9a	Merged revisions 330162 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r330162 \| pabelanger \| 2011-07-29 01:25:18 -0400 (Fri, 29 Jul 2011) \| 4 lines Fix typo pointed out on #asterisk Thanks notten ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330163 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	d4e239fa60	Merged revisions 329950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r329950 \| seanbright \| 2011-07-28 08:43:55 -0400 (Thu, 28 Jul 2011) \| 1 line Correct the spelling of 'conference.' ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	41630b37bc	Merged revisions 329538 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329538 \| jrose \| 2011-07-26 09:19:34 -0500 (Tue, 26 Jul 2011) \| 11 lines Merged revisions 329529 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329529 \| jrose \| 2011-07-26 09:04:55 -0500 (Tue, 26 Jul 2011) \| 5 lines Changes sound file for prepend "then-press-pound" to "vm-then-pound" which is the same prompt, only it turned out "then-press-pound" was part of extra sounds. Also, vm is more appropriate anyway. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329564 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	462e0fe530	Merged revisions 329528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329528 \| jrose \| 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) \| 24 lines Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 \| jrose \| 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) \| 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a2e30b1908	Merged revisions 329200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329200 \| rmudgett \| 2011-07-21 12:32:02 -0500 (Thu, 21 Jul 2011) \| 24 lines Merged revisions 329199 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329199 \| rmudgett \| 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011) \| 17 lines Update PickupChan documentation. The PickupChan uses the ampersand as the argument separator. Was documented as: PickupChan(channel[,channel2[,...][,options]]) Fixed documentation to: PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options]) This is a continuation of ASTERISK-17494 for v1.8 and later. (closes issue ASTERISK-18144) Reported by: Erik Smith Patches: pickupchan_ducumentation-v2.patch (License #6263) patch uploaded by Erik Smith Tested by: Erik Smith ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	4ea4b7e1ab	Merged revisions 328771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328771 \| kmoore \| 2011-07-19 10:46:54 -0500 (Tue, 19 Jul 2011) \| 18 lines Merged revisions 328770 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328770 \| kmoore \| 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) \| 11 lines MeetMe requests a PIN twice in some circumstances If a call to MeetMe includes both the dynamic(D) and always request PIN(P) options, MeetMe will ask for the PIN two times: once for creating the conference and once for entering the conference. This behavior was introduced in rev 311616 when adding the CONFFLAG_ALWAYSPROMPT option to the logic branch controlling PIN entry for joining a conference. (closes AST-601) Review: https://reviewboard.asterisk.org/r/1305/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328772 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Murawki	3719ee2d65	Merged revisions 328664 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328664 \| markm \| 2011-07-18 16:50:13 -0400 (Mon, 18 Jul 2011) \| 15 lines Merged revisions 328663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328663 \| markm \| 2011-07-18 16:47:04 -0400 (Mon, 18 Jul 2011) \| 9 lines app_dial may double free a channel datastore When starting a call with originate, and having the callee channel run Bridge() on pickup, we will double free the dialed_interface_info datastore, causing a crash. Make sure to check if the datastore still exists before trying to free it. (closes issue ASTERISK-17917) Reported by: Mark Murawski Tested by: Mark Murawski ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	37508c1946	Merged revisions 328451 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328451 \| lmadsen \| 2011-07-15 16:17:25 -0500 (Fri, 15 Jul 2011) \| 1 line Build app_macro by default because things depend on it. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328459 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	145c174565	Merged revisions 328329 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328329 \| rmudgett \| 2011-07-14 19:19:32 -0500 (Thu, 14 Jul 2011) \| 2 lines Make hint watcher callback take const strings for context and exten parameters. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	a525edea59	Merged revisions 328247 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 \| lmadsen \| 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) \| 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 \| lmadsen \| 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) \| 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	c0dc1ddb45	Merged revisions 328120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328120 \| dvossel \| 2011-07-13 17:09:34 -0500 (Wed, 13 Jul 2011) \| 15 lines Preserve sample rate quality of wideband mixmonitor recordings. MixMonitor has the ability to record in any file format Asterisk supports, but the quality of wideband audio is not preserved. This is because regardless of the sample rate the call is being recorded in, the audio is always downsampled to 8khz and then upsampled to whatever wideband format it is being written as. This patch resolves this by requesting the audio from the audiohook in the signed linear format closest to the sample rate of the format we are writing. This fix is only possible for Asterisk 1.10 because audio hooks in 1.8 are not capable of wideband audio. Review: https://reviewboard.asterisk.org/r/1314/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	ae3d614ab8	Merged revisions 327890 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327890 \| mnicholson \| 2011-07-12 15:07:20 -0500 (Tue, 12 Jul 2011) \| 2 lines search in the current context for 'a' and 'o' instead of 'default' ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327891 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	0fc745aaf1	Merged revisions 327852 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327852 \| mjordan \| 2011-07-12 14:10:34 -0500 (Tue, 12 Jul 2011) \| 12 lines Added additional checks for mailbox / password beginning with '' character A bug existed such that if a user entered a password with '', and the extension 'a' did not exist, an invalid mailbox would be created and the user authenticated. The code was changed to prevent this from occurring, and to prevent users from having mailboxes or passwords defined that begin with the '*' character. (closes issue ASTERISK-17443) Reported by: Kevin Scott Adams Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1316/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	934cabb22f	Segfault on shutdown when confbridge is active When undergoing a shutdown and channels are kicked out of a bridge, a segfault occurs because ConfBridge tries to play sounds on the bridge after the underlying channels have been blown away due to the shutdown. (closes ASTERISK-18040) Review: https://reviewboard.asterisk.org/r/1283/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327748 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	17860b70e4	Updates confbridge.conf video documentation and adds dtmf action for releasing video src. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326782 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	7d179abfd4	Merged revisions 326411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326411 \| tilghman \| 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) \| 14 lines 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/trunk@326412 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	1339a0a535	Video support for ConfBridge. Review: https://reviewboard.asterisk.org/r/1288/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	c81556d8ef	Merged revisions 325877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325877 \| mjordan \| 2011-06-30 15:09:48 -0500 (Thu, 30 Jun 2011) \| 9 lines 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/trunk@325900 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4240017462	Merged revisions 325614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325614 \| rmudgett \| 2011-06-29 13:16:45 -0500 (Wed, 29 Jun 2011) \| 5 lines 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/trunk@325616 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	54763625c6	Merged revisions 325610 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325610 \| rmudgett \| 2011-06-29 13:05:15 -0500 (Wed, 29 Jun 2011) \| 18 lines 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/trunk@325611 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	6c7d437287	Merged revisions 325537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325537 \| mnicholson \| 2011-06-29 10:34:47 -0500 (Wed, 29 Jun 2011) \| 2 lines don't do native/remote bridging if a framehook is active on the channel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	f99a06d030	Commit "distrotech" app_queue changes to Trunk	14 years ago
Kinsey Moore	67d4d6b656	ConfBridge: redundant code cleanup There is no reason to clean up features twice. Review: https://reviewboard.asterisk.org/r/1279/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	698bc02570	Fixes issue with channel write format being incorrectly restored when MOH is used in confbridge. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324422 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	1573ad78d2	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/trunk@324304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	a5770c43f0	Merged revisions 324176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324176 \| lmadsen \| 2011-06-17 14:38:40 -0400 (Fri, 17 Jun 2011) \| 2 lines Fix typo in documentation. Pointed out by Vlad Povorozniuc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	b019f95642	CONFBRIDGE_INFO function to get conference data Added the CONFBRIDGE_INFO dialplan function to get information about a conference bridge including locked status and number of parties, admins, and marked users. Review: https://reviewboard.asterisk.org/r/1271/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	40ea500078	Config inheritance doesn't work with ConfBridge() menu definitions Current behavior in ConfBridge menu definitions is that first definition takes precedence, even in templated situations. This change allows inheritance and overriding to work as expected so that the last definition takes precedence. (closes ASTERISK-17986) Review: https://reviewboard.asterisk.org/r/1267/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323272 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	42cb4cf514	MOH for only user not working with ConfBridge This adds the playing_moh flag to the conference_bridge_user struct that signifies when MOH should be playing so code doesn't have to guess whether MOH is playing. This change also adds the necessary checking to ensure that MOH continues playing for a single user in a conference after the join sound is played when configured to do so. (closes ASTERISK-17988) Review: https://reviewboard.asterisk.org/r/1263/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323107 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	cd15477923	ConfBridge: Use of bridge or user profiles that don't exist Bridge and user profiles are not checked for existence before use. The lack of a fully formed bridge profile can cause a segfault when sounds are accessed. This change ensures that bridge and user profiles exist prior to usage attempts. Review: https://reviewboard.asterisk.org/r/1264/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323106 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0a8f9d2cf0	Merged revisions 322749 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322749 \| rmudgett \| 2011-06-09 11:31:53 -0500 (Thu, 09 Jun 2011) \| 15 lines 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/trunk@322750 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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