23906 Commits (154fbf8cae866269cb2b64a5cb26cb4d435cf4df)

 

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Author	SHA1	Message	Date
Jason Parker	154fbf8cae	Split Hold event into Hold/Unhold, and move it into core. ... (closes issue ASTERISK-21487) Review: https://reviewboard.asterisk.org/r/2565/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389746 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	1223199b3d	Remove a junk define ... BLOB_HANDLER_BUCKETS is a remnant of using "type" fields in JSON/snapshot blobs and is no longer used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	06be8463b6	Migrate a large number of AMI events over to Stasis-Core ... This patch moves a number of AMI events over to the Stasis-Core message bus. This includes: * ChanSpyStart/Stop * MonitorStart/Stop * MusicOnHoldStart/Stop * FullyBooted/Reload * All Voicemail/MWI related events In addition, it adds some Stasis-Core and AMI support for generic AMI messages, refactors the message router in AMI to use a single router with topic forwarding for the topics that AMI cares about, and refactors MWI message types and topics to be more name compliant. Review: https://reviewboard.asterisk.org/r/2532 (closes issue ASTERISK-21462) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c1b51fd265	Print all logger messages on shutdown ... When Asterisk shuts down and shuts down the loggin gsubsystem, any messages currently in flight will not get logged. This patch prevents the loop writing messages from breaking out prematurely, such that all of the messages are logged. (closes issue ASTERISK-21716) Reported by: Corey Farrell patches: logger-process-all-messages.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 389676 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 389677 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	1fb6f365ec	Fix several problems caused by multiple line usage with i2004 phones. ...	12 years ago
David M. Lee	32a86f902a	stasis-http: Provide a response body for 201 created responses ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	9c7c1897e5	res_parking: Add a verbose message when a channel is parked ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	dd6923d0a9	res_parking: Fix some simple bugs ... Both of them are covered in the dynamic parking review on https://reviewboard.asterisk.org/r/2550 - Remove unref against parking lot that the bridge did on dissolve since the reference wasn't taken in the first place. On a swap, reapply bridge roles in order to get music on hold and such playing on the channel that swaps into the bridge. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	814fa7fe11	Fix a crash due to the INVITE session being destroyed before the session. ... This change ensures that the INVITE session remains valid for the lifetime of the session object itself by increasing the session count on the dialog that the INVITE session is allocated from. Once this reaches zero (normally as a result of decrementing it within the session destructor) the dialog, and INVITE session, are destroyed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389609 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	557125664d	This patch adds support for controlling a playback operation from the ... Asterisk REST interface. This adds the /playback/{playbackId}/control resource, which may be POSTed to to pause, unpause, reverse, forward or restart the media playback. Attempts to control a playback that is not currently playing will either return a 404 Not Found (because the playback object no longer exists) or a 409 Conflict (because the playback object is still in the queue to be played). This patch also adds skipms and offsetms parameters to the /channels/{channelId}/play resource. (closes issue ASTERISK-21587) Review: https://reviewboard.asterisk.org/r/2559 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	10ba6bf8a8	This patch implements the REST API's for POST /channels/{channelId}/play ... and GET /playback/{playbackId}. This allows an external application to initiate playback of a sound on a channel while the channel is in the Stasis application. /play commands are issued asynchronously, and return immediately with the URL of the associated /playback resource. Playback commands queue up, playing in succession. The /playback resource shows the state of a playback operation as enqueued, playing or complete. (Although the operation will only be in the 'complete' state for a very short time, since it is almost immediately freed up). (closes issue ASTERISK-21283) (closes issue ASTERISK-21586) Review: https://reviewboard.asterisk.org/r/2531/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3464e0919a	Fix inverted test preventing DTMF disconnect from working. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	61927f7c82	Fix a bug where the DTMF mode was not set on newly created RTP instances in the res_sip_sdp_rtp module. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389568 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	7f43acc49c	Fix a bug with applying the end result of the codec negotiation to the Asterisk channel. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	9d742946db	Fix a bug where the codec order as configured was not being obeyed. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	8a5a09e62c	Fixed startup race condition which caused occasional stasis_mwi_state_type assertions. ... The caching topic (which refers to the message type) was created before the message type. If the initial subscription message gets processed before the type can be initialized, the assertion about using an uninitialized type fires. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	1355e5d34f	Remove bad props, before anybody notices. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	b6aac885be	Add dial events to app_queue and app_followme. ... Also fixes an issue in app_dial, where the channels were swapped on dial events. (closes issue ASTERISK-21551) (closes issue ASTERISK-21550) Review: https://reviewboard.asterisk.org/r/2549/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389492 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	054efbc45a	Fix destruction order assert for stasis_bridging ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389454 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	908ac3507a	Conditional out more app_queue logging that needs to be reworked. ... Fixes crash because app_queue was unconditionally freeing a datastore that was still on a channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	afb1d96068	Raise the ConfBridgeMute/Unmute events when a CLI or AMI action triggers the change ... New in 12 are the ConfBridgeMute/Unmute events, which are triggered when a user changes their mute/unmute state. This was typically triggered when a user hit a DTMF key that triggered the mute/unmute menu handler. Forgotten in this is when an AMI action or CLI command triggers the mute/unmute. This patch now raises the events in those situations as well. (closes issue ASTERISK-21802) Reported by: Birger "WIMPy" Harzenetter git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389402 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3d63833bd6	Merge in the bridge_construction branch to make the system use the Bridging API. ... Breaks many things until they can be reworked. A partial list: chan_agent chan_dahdi, chan_misdn, chan_iax2 native bridging app_queue COLP updates DTMF attended transfers Protocol attended transfers git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	e1e1cc2dee	Fixed some extra field assertion when the event WebSocket is connected ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	d8aec72494	Set the AST_CDR_FLAG_ORIGINATED flag on originated channel's CDRs ... This may alleviate some of the CDR woes with originated channels, as CDRs do like to know when a channel was originated. Eventually this will get converted to be a channel flag, so its location is still good to know post the great CDR shakeup of 2013. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389306 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d8f40ed39a	Fixup svn:keywords in all *.c and *.h files. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389251 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	81171a4f4b	Fixup svn:keywords in all *.c and *.h files. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	a042955a24	Add doxygen.log to svn:ignore property. ... ........ Merged revisions 389244 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 389245 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389246 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	e9a0ab42da	Add missing exports file ... This exposes stasis_app_control_answer and allows res_stasis_http_channels to load properly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	734a154eef	In Sorcery pass the name of the object being allocated to the allocator. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	eb06c505f4	Add documentation for record_file_append ... When this option was added, it was noted in CHANGES, but was missing the XML documentation that this patch adds. (closes issue ASTERISK-21780) Patch-by: Brad Latus (snuffy) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389202 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alexandr Anikin	62e64134fa	add ast_publish_channel_state according new event framework ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	8d6b0c9f3b	Add transfer softkey to ringout state to enable blond transfers. ... (closes issue ASTERISK-21327) Reported by: wedhorn Tested by: myself Patches: skinny-blindxfer01.diff uploaded by wedhorn (license 5019) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	1b5a3069f9	Add base XML documentation for res_sip ... Thanks to Brad Latus, this patch adds a significant amount much-needed documentation to res_sip. It should cover all existing configuration options currently in Asterisk trunk. Patch-by: Brad Latus (snuffy) Review: https://reviewboard.asterisk.org/r/2471/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	b46840ae3e	Don't hold the outgoing lock for a prolonged period of time as it may block the originator. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	4d8c35abf2	If the caller of the originate API calls wants the channel ensure it has been requested and dialed. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	01d6e8dbc9	Add call forward no answer to skinny and cleanup general callfwd handling. ... CallforwardNoAnswer uses a sched to determine when to forward the call. Defaults to 20secs but configurable in skinny.conf. Adds dialType to each subchannel structure to be used to differentiate between normal dials that result in a call being placed (default) and other uses for the skinny_dialer (such as cfwd digit collection). Restructured all cfwd handling to use this new arrangement. (closes issue ASTERISK-21292) Reported by: wedhorn Tested by: myself Patches: skinny-callfwdnoans03.diff uploaded by wedhorn (license 5019) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389097 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	7316abeb8f	Fix a bug where synchronous origination (oddly enough triggered by doing an async manager Originate) would not work properly. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389085 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	4e38a4eb64	Move origination to use the dialing API and send Stasis messages on dial begin and end. ... (closes issue ASTERISK-21549) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2512/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389053 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	b97c71bb11	Fix shutdown assertions in stasis-core ... In r388005, macros were introduced to consistently define message types. This added an assert if a message type was used either before it was initialized or after it had been cleaned up. It turns out that this assertion fires during shutdown. This actually exposed a hidden shutdown ordering problem. Since unsubscribing is asynchronous, it's possible that the message types used by the subscription could be freed before the final message of the subscription was processed. This patch adds stasis_subscription_join(), which blocks until the last message has been processed by the subscription. Since joining was most commonly done right after an unsubscribe, a stasis_unsubscribe_and_join() convenience function was also added. Similar functions were also added to the stasis_caching_topic and stasis_message_router, since they wrap subscriptions and have similar problems. Other code in trunk was refactored to join() where appropriate, or at least verify that the subscription was complete before being destroyed. Review: https://reviewboard.asterisk.org/r/2540 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	91bab76422	Remove Character Limit On "inkeys" For IAX2 ... Currently, the buffer for processing "inkeys" is limited to 256 characters. If the user has many keys and the names of those key files are long, the 256 character limit is not enough. * Change inkeys buffer to be dynamic (closes issue ASTERISK-21398) Reported by: Pavel Kopchyk Tested by: Pavel Kopchyk, Michael L. Young Patches: asterisk-21398-iax2-inkeys-dynamic-buffer_v3.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2501/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	d04f1fd60a	Publish the outbound channel's application/data when dialing ... This patch does two things: * It fixes a bug where the outbound channel's application/data set by the dialing API/app_dial is not communicated until the channel is hung up. If that happens, AMI would incorrectly send a NewExten event immediately after a Hangup. This isn't really AMI's fault, as the dialing APIs never communicated the 'helpful' app/data on the outbound channel until it was hungup. * It makes public sending a stasis message about a change in channel state. This is useful enough that - for now at least - it should be public. If operations on a channel go to being more coarse-grained, this function could be made private again. Review: https://reviewboard.asterisk.org/r/2548 Note that this problem was found and reported by Matt DiMeo. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388976 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	b90bba7a30	Stasis: Update security events to use Stasis ... Also moves ACL messages to the security topic and gets rid of the ACL topic (closes issue ASTERISK-21103) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2496/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	15945a7185	Fixed inverted logic in app_add_channel(). ... Also added some missing doc comments for stasis/app.h. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	2eebab3992	Fix for segfault in __ast_rwlock_destroy with DEBUG_THREADS ... If DEBUG_THREADS is enabled __ast_rwlock_destroy causes a segfault while trying to access a possible NULL t->track object. A NULL check has been added before trying to access the memory. (closes issue ASTERISK-21724) Reported by: Corey Farrell Fixed by: Corey Farrell Patches: ast_rwlock_destroy-segv.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 388838 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 388839 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	d8d1def22c	Fix VM snapshot handling for combined INBOX. ... The snapshot API contains an option that allow for combining of new and old messages within a single snapshot. New messages, however, include options beyond just 'INBOX' - it also includes the Urgent folder. A previous patch that combined INBOX and Urgent accidentally impacted snapshots that attempted to gain messages from just the Old folder. This patch fixes the snapshot gathering such that the API returns the appropriate messages for the folder selected, with and without the combine option. This should make it more clear about what's happening. Review: https://reviewboard.asterisk.org/r/2539/ ........ Merged revisions 388816 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388818 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	1ead1853f2	Use srtp_shutdown when available ... This allows the SRTP library to be shut down properly when the functionality is offered by libsrtp. Review: https://reviewboard.asterisk.org/r/2538/ (closes issue ASTERISK-21719) ........ Merged revisions 388768 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 388769 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	9648e258c7	Refactored the rest of the message types to use the STASIS_MESSAGE_TYPE_* ... macros. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388751 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	e8f4ac6c61	Break res_stasis into smaller files. ... When implementing playback for stasis-http, the monolithicedness of res_stasis really started to get in my way. This patch breaks the major components of res_stasis.c into individual files. * res/stasis/app.c - Stasis application tracking * res/stasis/control.c - Channel control objects * res/stasis/command.c - Channel command object This refactoring also allows res_stasis applications to be loaded as independent modules, such as the new res_stasis_answer module. The bulk of this patch is simply moving code from one file to another, adjusting names and adding accessors as necessary. Review: https://reviewboard.asterisk.org/r/2530/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d1d1425327	Make ao2 global objects not always use the debug version of the ao2_ref() calls. ... The debug versions of ao2_ref() should only be used if REF_DEBUG is enabled so nothing is written to /tmp/refs unexpectedly. (closes issue ASTERISK-21785) Reported by: abelbeck Patches: jira_asterisk_21785_v11.patch (license #5621) patch uploaded by rmudgett Tested by: abelbeck ........ Merged revisions 388700 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	2a9cbd693e	Move JSON event generators into separate modules ... This moves the JSON event generators out of the Stasis-HTTP modules and into standalone JSON-related counterparts so that Stasis-HTTP and res_stasis can depend on them without creating dependency cycles. This also provides a future location for Swagger Model validator functions once the generators for that code are written. Review: https://reviewboard.asterisk.org/r/2534/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388668 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago