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2013-06-07Implement ARI POST to /channels, to originate a call.Jason Parker
(closes issue ASTERISK-21617) Review: https://reviewboard.asterisk.org/r/2597/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390885 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-07res_parking: Automatically generate extensions, hints, etc.Jonathan Rose
(closes issue ASTERISK-21645) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2545/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390849 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-06Fix documentation generationKinsey Moore
Regression from r390701 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-06Fix documentation that was in review during the great suffix/prefix swapKinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390728 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-06Split AGI manager events, to remove SubEvent field.Jason Parker
This moves them to stasis, in the process. (closes issue ASTERISK-21470) Review: https://reviewboard.asterisk.org/r/2587/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-05Remove remaining traces of remove_on_pull from hooks and hook APIs.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390550 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-29Pack the right number of items into the status and receive fax blobsMatthew Jordan
The code was still attempting to pack an additional item into the blobs that didn't exist. Crashes ensued. This patch modifies the publishing of these messages so that the correct number of items are packed in the JSON. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-29Resolve a merge conflictKinsey Moore
When ast_channel_cached_blob_create was merged, ast_channel_blob_create_from_cache was partially removed in an unresolved merge conflict. This restores ast_channel_blob_create_from_cache and refactors usage of ast_channel_cached_blob_create (requires an ast_channel) to use ast_channel_blob_create_from_cache (requires a channel uniqueid) instead. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-27Fix some more fax test errors due to needing the peer in a bridgeMatthew Jordan
In r389799, a number of fax errors in gateway mode were fixed by using the appropriate function to get a channel's peer while in a bridge. This patch does two things: (1) It uses the same function in res_fax_spandsp while starting the fax gateway. Without this, the fax gateway will not actually start up, as res_fax_spandsp also must inspect the channel's peer in a two-party bridge (2) It refactors some ao2 objects in sendfax_exec to use RAII_VAR. This was reverted in r389799 as some off nominal paths were getting hit without the fix in (1) that indicated an ao2 object issue; this turned out to be a red herring (which is an odd phrase) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-26Fix a few fax gateway failuresMatthew Jordan
Fax gateway requires knowledge of a channel's peer in a bridge. This patch now uses the supported mechanisms to get this information. This is acceptable for a few reasons: * Fax gateway can only ever work in a 2-party bridge * Fax gateway cannot work when not in a bridge * Fax gateway cannot work without knowledge of the capabilities of both channels in the fax operation (it is, after all, a gateway) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-26Fix a variety of memory corruption/assertion errorsMatthew Jordan
* Initialize a Stasis-Core message type prior to initializing a caching topic. The caching topic will attempt to use the message type. * Don't attempt to publish Stasis-Core messages from remote console connections. They aren't the main process; they shouldn't attempt to behave as it (they also don't have the infrastructure to do so) * Don't treat a JSON object as an ao2 object (whoops) * In asterisk.c, ref bump the JSON even package that is distributed with the event meta data. The callers assume that they own the reference, and the packing routine steals references. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-24Split Hold event into Hold/Unhold, and move it into core.Jason Parker
(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
2013-05-24Remove a junk defineKinsey Moore
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
2013-05-24Migrate a large number of AMI events over to Stasis-CoreMatthew Jordan
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
2013-05-23stasis-http: Provide a response body for 201 created responsesDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-23res_parking: Add a verbose message when a channel is parkedJonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-23res_parking: Fix some simple bugsJonathan Rose
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
2013-05-23Fix a crash due to the INVITE session being destroyed before the session.Joshua Colp
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
2013-05-23This patch adds support for controlling a playback operation from theDavid M. Lee
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
2013-05-23This patch implements the REST API's for POST /channels/{channelId}/playDavid M. Lee
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
2013-05-23Fix a bug where the DTMF mode was not set on newly created RTP instances in ↵Joshua Colp
the res_sip_sdp_rtp module. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-23Fix a bug with applying the end result of the codec negotiation to the ↵Joshua Colp
Asterisk channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-23Fix a bug where the codec order as configured was not being obeyed.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-21Merge in the bridge_construction branch to make the system use the Bridging API.Richard Mudgett
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
2013-05-20Add missing exports fileKinsey Moore
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
2013-05-19Add base XML documentation for res_sipKinsey Moore
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
2013-05-17Fix shutdown assertions in stasis-coreDavid M. Lee
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
2013-05-17Stasis: Update security events to use StasisJonathan Rose
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
2013-05-15Fixed inverted logic in app_add_channel().David M. Lee
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
2013-05-15Use srtp_shutdown when availableKinsey Moore
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
2013-05-15Refactored the rest of the message types to use the STASIS_MESSAGE_TYPE_*David M. Lee
macros. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-14Break res_stasis into smaller files.David M. Lee
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
2013-05-14Move JSON event generators into separate modulesKinsey Moore
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
2013-05-13Revert r388529 for nowKinsey Moore
Adding the cleanup function needs some deeper thought since it apparently doesn't exist for all variants of libsrtp. ........ Merged revisions 388596 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 388597 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-13Close libsrtp properlyKinsey Moore
Ensure that libsrtp is shutdown properly when res_srtp is unloaded. (closes issue ASTERISK-21719) Reported by: Corey Farrell Patches: res_srtp-library-shutdown.patch uploaded by Corey Farrell ........ Merged revisions 388529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 388530 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-10Fix Finding Extensions With Patterns Using ODBC RealtimeMichael L. Young
After the merge of support for the realtime sorcery module, extensions that contained a pattern were not being found through odbc realtime. It was tracked down to this one line that was advancing to the next variable list before it should have been. The removal of this one line fixes this. Tested this fix on my machine. Received confirmation that this is the right fix from file on IRC. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-10Address unload order issues for res_stasis* modulesDavid M. Lee
I've noticed when doing a graceful shutdown that the res_stasis_http.so module gets unloaded before the modules that use it, which causes some asserts during their unload. While r386928 was a quick hack to get it to not assert and die, this patch increases the use counts on res_stasis.so and res_stasis_http.so properly. It's a bigger change than I expected, hence the review instead of just committing it. Review: https://reviewboard.asterisk.org/r/2489/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-10Add channel events for res_stasis appsKinsey Moore
This change adds a framework in res_stasis for handling events from channel topics. JSON event generation and validation code is created from event documentation in rest-api/api-docs/events.json to assist in JSON event generation, ensure consistency, and ensure that accurate documentation is available for ALL events that are received by res_stasis applications. The userevent application has been refactored along with the code that handles userevent channel blob events to pass the headers as key/value pairs in the JSON blob. As a side-effect, app_userevent now handles duplicate keys by overwriting the previous value. Review: https://reviewboard.asterisk.org/r/2428/ (closes issue ASTERISK-21180) Patch-By: Kinsey Moore <kmoore@digium.com> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-09Fix The Payload Being Set On CN Packets And Do Not Set Marker BitMichael L. Young
When we send out a CN packet (for instance, in the case of using rtpkeepalives), we are not setting the payload code properly. Also, we are setting the marker bit when we shouldn't be according to RFC 3389, section 4. AST_RTP_CN is not defined by AST_FORMAT codes. Therefore, we should be using ast_rtp_codecs_payload_code() rather than ast_rtp_codecs_payload_lookup(). 11 and trunk already use the appropriate function. * In 1.8, use ast_rtp_codecs_payload_code() * Remove the setting of the marker bit * Fix the debug message by incrementing the seqno after the debug message is set in order to display the correct seqno that was sent out (closes issue ASTERISK-21246) Reported by: Peter Katzmann Tested by: Peter Katzmann, Michael L. Young Patches: asterisk-21246-rtp-cng-payload-error_1.8_v2.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2500/ ........ Merged revisions 388111 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 388112 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388113 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-08Fixed MODFLAG for res_stasis_websocketDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-08Remove required type field from channel blobsDavid M. Lee
When we first introduced the channel blob types, the JSON blobs were self identifying by a required "type" field in the JSON object itself. This, as it turns out, was a bad idea. When we introduced the message router, it was useless for routing based on the JSON type. And messages had two type fields to check: the stasis_message_type() of the message itself, plus the type field in the JSON blob (but only if it was a blob message). This patch corrects that mistake by removing the required type field from JSON blobs, and introducing first class stasis_message_type objects for the actual message type. Since we now will have a proliferation of message types, I introduced a few macros to help reduce the amount of boilerplate necessary to set them up. Review: https://reviewboard.asterisk.org/r/2509 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-08Initial support for endpoints.David M. Lee
An endpoint is an external device/system that may offer/accept channels to/from Asterisk. While this is a very useful concept for end users, it is surprisingly not a core concept within Asterisk itself. This patch defines ast_endpoint as a separate object, which channel drivers may use to expose their concept of an endpoint. As the channel driver creates channels, it can use ast_endpoint_add_channel() to associate channels to the endpoint. This updated the endpoint appropriately, and forwards all of the channel's events to the endpoint's topic. In order to avoid excessive locking on the endpoint object itself, the mutable state is not accessible via getters. Instead, you can create a snapshot using ast_endpoint_snapshot_create() to get a consistent snapshot of the internal state. This patch also includes a set of topics and messages associated with endpoints, and implementations of the endpoint-related RESTful API. chan_sip was updated to create endpoints with SIP peers, but the state of the endpoints is not updated with the state of the peer. Along for the ride in this patch is a Stasis test API. This is a stasis_message_sink object, which can be subscribed to a Stasis topic. It has functions for blocking while waiting for conditions in the message sink to be fulfilled. (closes issue ASTERISK-21421) Review: https://reviewboard.asterisk.org/r/2492/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-07Minor fixups to Doxygen comments.David M. Lee
The \example tags marks an entire file as an example, not a code snippet. ........ Merged revisions 387823 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387824 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-06Add support for observers and JSON objectset creation to sorcery.Joshua Colp
This change adds the ability for modules to add themselves as observers to sorcery object types. Observers can be notified when objects are created, updated, or deleted as well as when the object type is loaded or reloaded. Observer notifications are done using a thread pool in a serialized fashion so the caller of the sorcery API calls is minimally impacted. This also adds the ability to create JSON changesets of a sorcery object. Tests are also present to confirm all of the above functionality. Review: https://reviewboard.asterisk.org/r/2477/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-03Stasis: Convert network change events into network change stasis messagesJonathan Rose
(issue ASTERISK-21103) Review: https://reviewboard.asterisk.org/r/2490/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-03Use the configured formats for Gulp sessions if there are no joint formats ↵Joshua Colp
between requested formats and configured formats. (closes issue ASTERISK-21756) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-01Clear the DTMF sending digit tracking on off nominal pathsMatthew Jordan
In certain situations, when the RTP engine goes to send a DTMF end digit it may be in a situation where the remote address is no longer available, or the digit that was supposed to be sent is invalid. In such cases, we need to clear the RTP counters appropriately. Otherwise, when the RTP source is set again, we'll continue to think that we're in the middle of sending a DTMF digit, which can confuse the remote party (signficantly). (closes issue ASTERISK-21522) Reported by: Corey Farrell patches: rtp_dtmf_process_end.patch uploaded by Corey Farrell (License 5909) ........ Merged revisions 387213 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 387216 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-30Just a couple of Stasis-HTTP nitpick fixes.David M. Lee
* Fixed crash when res_stasis_http is unloaded before the implementation modules. * Cleaned up test initialization for test_stasis_http.so. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-27Tweak res_sip priority so it gets loaded first before all other SIP stuff.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-27Update res_config_sqlite to use the ast_variable lists.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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