mirror of
https://github.com/asterisk/asterisk.git
synced 2025-10-27 14:41:58 +00:00
05d05e68f5
------------------------------------------------------------------------ r365395 | qwell | 2012-05-04 16:17:08 -0500 (Fri, 04 May 2012) | 7 lines Add support for folders in MixMonitor 'm' option. Backport manager actions. The manager actions are needed, so MixMonitor can be executed on existing channels. (issue DPMA-68) ------------------------------------------------------------------------ r364761 | qwell | 2012-05-01 12:25:14 -0500 (Tue, 01 May 2012) | 6 lines Remove folder_dir from voicemail snapshots API. It was both unused (except in tests, where it was fudged) and unnecessary. (closes issue AST-842) ------------------------------------------------------------------------ r367161 | mmichelson | 2012-05-21 14:05:52 -0500 (Mon, 21 May 2012) | 21 lines Add "send to voicemail" Digium phone functionality to Asterisk. This change accommodates two methods by which calls can be directed to a user's voicemail. * Incoming calls can be redirected to any user's voicemail. * Established calls can be blind transferred to any user's voicemail. Digium phones indicate the desire to direct a call to voicemail by using a Diversion header with a reason parameter of "send_to_vm". This patch adds the "send_to_vm" reason as a valid redirecting reason. In addition, chan_sip.c has been modified to update redirecting information on the transferred channel by reading a Diversion header on a REFER request. (closes issue AST-871) Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/1925 ------------------------------------------------------------------------ r368790 | mjordan | 2012-06-12 08:44:36 -0500 (Tue, 12 Jun 2012) | 18 lines Fix deadlock in SIP transfers that involve a REFER request In r367163, "send to voicemail" functionality was added to the SIP channel driver. This required updating the party redirecting information for the channel based on the headers provided in the REFER request. When the redirecting party information is updated on the channel, a call to ast_indicate_data occurs. Because handle_request_refer still had the sip_pvt locked, a deadlock could occur between the pbx_thread and the do_monitor thread servicing the REFER request. This patch preserves the proper locking order between the channel and the sip_pvt by ensuring that the sip_pvt is unlocked prior to updating the party redirecting information on the channel. (closes issue AST-903) Reported by: Matt Jordan patches: jira_ast_903_trunk.patch by rmudgett (license 5621) ------------------------------------------------------------------------ r368962 | qwell | 2012-06-14 13:38:48 -0500 (Thu, 14 Jun 2012) | 11 lines Remove global symbol requirement from app_voicemail. This uses the existing "function installation" stuff that already existed for other functions, like getting message counts. (closes issue AST-807) (issue AST-901) (issue AST-908) Review: https://reviewboard.asterisk.org/r/1965/ ------------------------------------------------------------------------ r368964 | qwell | 2012-06-14 14:03:24 -0500 (Thu, 14 Jun 2012) | 8 lines These functions that were moved need to be static. Also wrap test functions in a #ifdef. (issue AST-807) (issue AST-901) (issue AST-908) ------------------------------------------------------------------------ r368998 | qwell | 2012-06-15 10:31:43 -0500 (Fri, 15 Jun 2012) | 6 lines Remove some symbol exports that got missed in the removal of global symbols. (issue AST-807) (issue AST-901) (issue AST-908) ------------------------------------------------------------------------ r369024 | qwell | 2012-06-15 11:29:40 -0500 (Fri, 15 Jun 2012) | 2 lines Fix voicemail API tests by using the correct argument order for create/destroy. ------------------------------------------------------------------------ git-svn-id: https://origsvn.digium.com/svn/asterisk/certified/branches/1.8.11@369839 65c4cc65-6c06-0410-ace0-fbb531ad65f3