95 Commits (f080ca65367a7725889ffd9acb6179b9b150993d)

Author	SHA1	Message	Date
Matthew Jordan	fb8a2e0399	ARI: Improve wiki documentation ... This patch improves the documentation of ARI on the wiki. Specifically, it addresses the following: * Allowed values and allowed ranges weren't documented. This was particularly frustrating, as Asterisk would reject query parameters with disallowed values - but we didn't tell anyone what the allowed values were. * The /play/id operation on /channels and /bridges failed to document all of the added media resource types. * Documentation for creating a channel into a Stasis application failed to note when it occurred, and that creating a channel into Stasis conflicts with creating a channel into the dialplan. * Some other minor tweaks in the mustache templates, including italicizing the parameter type, putting the default value on its own sub-bullet, and some other nicities. Review: https://reviewboard.asterisk.org/r/4351 ........ Merged revisions 431145 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Ashley Sanders	804ab70f9d	ARI: Fixed crash that occurred when updating a bridge when the optional query parameter 'name' was not supplied. ... Prior to this changeset, posting to the: /ari/bridges/{bridgeId} endpoint without specifying a value for the [name] query parameter, would crash Asterisk if the bridge you are attempting to create (or update) had the same ID as an existing bridge. The internal mechanism of the POST operation interpreted a null value for name, thus resulting in an error condition that crashed Asterisk. ASTERISK-24560 #close Reported By: Kinsey Moore Review: https://reviewboard.asterisk.org/r/4349/ ........ Merged revisions 430818 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430820 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	7f836c1c15	Add the ability to continue and originate using priority labels. ... With this patch, the following two ARI commands POST /channels POST /channels/{id}/continue Accept a new parameter, label, that can be used to continue to or originate to a priority label in the dialplan. Because this is adding a new parameter to ARI commands, the API version of ARI has been bumped from 1.6.0 to 1.7.0. This patch comes courtesy of Nir Simionovich from Greenfield Tech. Thanks! ASTERISK-24412 #close Reported by Nir Simionovich Review: https://reviewboard.asterisk.org/r/4285 ........ Merged revisions 430337 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	d673209abc	ARI/AMI: Include language in standard channel snapshot output ... The channel "language" was already part of a channel snapshot, however is was not sent out over AMI or ARI. This patch makes it so the channel "language" is included in the appropriate AMI or ARI events. ASTERISK-24553 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4245/ ........ Merged revisions 429204 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 429206 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	60ab564ad2	ari: Add support for specifying an originator channel when originating. ... If an originator channel is specified when originating a channel the linked ID of it will be applied to the newly originated outgoing channel. This allows an association to be made between the two so it is known that the originator has dialed the originated channel. ASTERISK-24552 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4243/ ........ Merged revisions 429153 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	fe7671fee6	Add new AMI and ARI events for connected line changes on a channel. ... The AMI event is called NewConnectedLine and the ARI event is called ChannelConnectedLine. ASTERISK-24554 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/4231 ........ Merged revisions 429064 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	36f4bff943	Stasis: Add information to blind transfer event ... When a blind transfer occurs that is forced to create a local channel pair to satisfy the transfer request, information about the local channel pair is not published. This adds a field to describe that channel to the blind transfer message struct so that this information is conveyed properly to consumers of the blind transfer message. This also fixes a bug in which Stasis() was unable to properly identify the channel that was replacing an existing Stasis-controlled channel due to a blind transfer. Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3921/ ........ Merged revisions 421537 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421538 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	bb494067a5	Multiple revisions 421311-421312 ... ........ r421311 | mjordan | 2014-08-17 20:11:28 -0500 (Sun, 17 Aug 2014) | 9 lines res/ari/resource_channels: Don't return allocation failure on failed function If a function fails to execute, it is most likely due to one of two reasons: (1) The function doesn't exist or can't be read from (2) The function is dangerous and is restricted based on the user's permissions Currently we return allocation failure, which is incorrect. This updates the reason code to more accurately reflect why the request failed. ASTERISK-24215 ........ r421312 | mjordan | 2014-08-17 20:13:41 -0500 (Sun, 17 Aug 2014) | 4 lines res/ari/resource_channels: Fix compilation issue Forgot a parameter. Whoops. ........ Merged revisions 421311-421312 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ba5d5da60b	Improve call forwarding reporting, especially with regards to ARI. ... This patch addresses a few issues: 1) The order of Dial events have been changed when performing a call forward. The order has now been altered to 1) Dial begins dialing channel A. 2) When A forwards the call to B, we issue the dial end event to channel A, indicating the dial is being canceled due to a forward to B. 3) When the call to channel B occurs, we then issue a new dial begin to channel B. 2) Call forwards are now reported on the calling channel, not the peer channel. 3) AMI DialEnd events have been altered to display the extension the call is being forwarded to when relevant. 4) You can now get the values of channel variables for channels that are not currently in the Stasis application. This brings the retrieval of channel variables more in line with the rest of channel read operations since they may be performed on channels not in Stasis. ASTERISK-24134 #close Reported by Matt Jordan ASTERISK-24138 #close Reported by Matt Jordan Patches: forward-shenanigans.diff uploaded by Matt Jordan (License #6283) Review: https://reviewboard.asterisk.org/r/3899 ........ Merged revisions 420794 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	79c5c08db9	ARI: Originate to app local channel subscription code optimization. ... Reduce the scope of local_peer and only get it if the ARI originate is subscribing to the channels. Review: https://reviewboard.asterisk.org/r/3905/ ........ Merged revisions 421009 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421010 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421012 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	6f735288b0	Stasis: Allow internal channels directly into bridges ... The patch to catch channels being shoehorned into Stasis() via external mechanisms also happens to catch Announcer and Recorder channels because they aren't known to be stasis-controlled channels in the usual sense. This marks those channels as Stasis()-internal channels and allows them directly into bridges. Review: https://reviewboard.asterisk.org/r/3903/ ........ Merged revisions 420795 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 420796 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420797 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	0ac7f96057	Stasis: Convey transfer information to applications ... This fixes a class of issues where Stasis applications were not made aware that their channels were being manipulated or replaced by external entitiessuch as transfers, AMI commands, or dialplan applications such as Bridge(). Inconsistent information such as StasisEnd events with unknown channels as a result of masquerades has also been corrected. To accomplish these fixes, several new fields were added to blind and attended transfer messages as well as StasisStart and BridgeAttendedTransfer Stasis events. ASTERISK-23941 #close Review: https://reviewboard.asterisk.org/r/3865/ Review: https://reviewboard.asterisk.org/r/3857/ Review: https://reviewboard.asterisk.org/r/3852/ Review: https://reviewboard.asterisk.org/r/3816/ Review: https://reviewboard.asterisk.org/r/3731/ Review: https://reviewboard.asterisk.org/r/3729/ Review: https://reviewboard.asterisk.org/r/3728/ ........ Merged revisions 420325 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	47bf7efc4d	Multiple revisions 420089-420090,420097 ... ........ r420089 | mjordan | 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) | 72 lines ARI: Add channel technology agnostic out of call text messaging This patch adds the ability to send and receive text messages from various technology stacks in Asterisk through ARI. This includes chan_sip (sip), res_pjsip_messaging (pjsip), and res_xmpp (xmpp). Messages are sent using the endpoints resource, and can be sent directly through that resource, or to a particular endpoint. For example, the following would send the message "Hello there" to PJSIP endpoint alice with a display URI of sip:asterisk@mycooldomain.org: ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There This is equivalent to the following as well: ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There Both forms are available for message technologies that allow for arbitrary destinations, such as chan_sip. Inbound messages can now be received over ARI as well. An ARI application that subscribes to endpoints will receive messages from those endpoints: { "type": "TextMessageReceived", "timestamp": "2014-07-12T22:53:13.494-0500", "endpoint": { "technology": "PJSIP", "resource": "alice", "state": "online", "channel_ids": [] }, "message": { "from": "\"alice\" <sip:alice@127.0.0.1>", "to": "pjsip:asterisk@127.0.0.1", "body": "Watson, come here.", "variables": [] }, "application": "testsuite" } The above was made possible due to some rather major changes in the message core. This includes (but is not limited to): - Users of the message API can now register message handlers. A handler has two callbacks: one to determine if the handler has a destination for the message, and another to handle it. - All dialplan functionality of handling a message was moved into a message handler provided by the message API. - Messages can now have the technology/endpoint associated with them. Various other properties are also now more easily accessible. - A number of ao2 containers that weren't really needed were replaced with vectors. Iteration over ao2_containers is expensive and pointless when the lifetime of things is well defined and the number of things is very small. res_stasis now has a new file that makes up its structure, messaging. The messaging functionality implements a message handler, and passes received messages that match an interested endpoint over to the app for processing. Note that inadvertently while testing this, I reproduced ASTERISK-23969. res_pjsip_messaging was incorrectly parsing out the 'to' field, such that arbitrary SIP URIs mangled the endpoint lookup. This patch includes the fix for that as well. Review: https://reviewboard.asterisk.org/r/3726 ASTERISK-23692 #close Reported by: Matt Jordan ASTERISK-23969 #close Reported by: Andrew Nagy ........ r420090 | mjordan | 2014-08-05 15:16:37 -0500 (Tue, 05 Aug 2014) | 2 lines Remove automerge properties :-( ........ r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue, 05 Aug 2014) | 2 lines test_message: Fix strict-aliasing compilation issue ........ Merged revisions 420089-420090,420097 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	355dc3d2ad	Multiple revisions 419565-419566 ... ........ r419565 | mjordan | 2014-07-25 09:41:23 -0500 (Fri, 25 Jul 2014) | 21 lines ARI: report duration values in LiveRecording objects This patch adds three new fields to the LiveRecording model: - total_duration: the total length of the live recording - talking_duration: optional. The duration of talking energy that was detected while the recording was made. - silence_duration: optional. The duration of silence that was detected while the recording was made. These values are reported in the RecordingFinished ARI event. When a DSP is enabled on the channel during the recording - which occurs when the recording is created with max_silence_seconds (indicating that the user actually cares about how much silence is in the file), we will report the talking_duration and silence_duration in addition to the total_duration. Review: https://reviewboard.asterisk.org/r/3770/ ASTERISK-24037 #close Reported by: Samuel Galarneau ........ r419566 | mjordan | 2014-07-25 09:46:15 -0500 (Fri, 25 Jul 2014) | 1 line Update CHANGES for r419565 ........ Merged revisions 419565-419566 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	bb87796f67	ARI: Fix endpoint/channel subscription issues; allow for subscriptions to tech ... This patch serves two purposes: (1) It fixes some bugs with endpoint subscriptions not reporting all of the channel events (2) It serves as the preliminary work needed for ASTERISK-23692, which allows for sending/receiving arbitrary out of call text messages through ARI in a technology agnostic fashion. The messaging functionality described on ASTERISK-23692 requires two things: (1) The ability to send/receive messages associated with an endpoint. This is relatively straight forwards with the endpoint core in Asterisk now. (2) The ability to send/receive messages associated with a technology and an arbitrary technology defined URI. This is less straight forward, as endpoints are formed from a tech + resource pair. We don't have a mechanism to note that a technology that *may* have endpoints exists. This patch provides such a mechanism, and fixes a few bugs along the way. The first major bug this patch fixes is the forwarding of channel messages to their respective endpoints. Prior to this patch, there were two problems: (1) Channel caching messages weren't forwarded. Thus, the endpoints missed most of the interesting bits (such as channel creation, destruction, state changes, etc.) (2) Channels weren't associated with their endpoint until after creation. This resulted in endpoints missing the channel creation message, which limited the usefulness of the subscription in the first place (a major use case being 'tell me when this endpoint has a channel'). Unfortunately, this meant another parameter to ast_channel_alloc. Since not all channel technologies support an ast_endpoint, this patch makes such a call optional and opts for a new function, ast_channel_alloc_with_endpoint. When endpoints are created, they will implicitly create a technology endpoint for their technology (if one does not already exist). A technology endpoint is special in that it has no state, cannot have channels created for it, cannot be created explicitly, and cannot be destroyed except on shutdown. It does, however, have all messages from other endpoints in its technology forwarded to it. Combined with the bug fixes, we now have Stasis messages being properly forwarded. Consider the following scenario: two PJSIP endpoints (foo and bar), where bar has a single channel associated with it and foo has two channels associated with it. The messages would be forwarded as follows: channel PJSIP/foo-1 -- \ --> endpoint PJSIP/foo -- / \ channel PJSIP/foo-2 -- \ ---- > endpoint PJSIP / channel PJSIP/bar-1 -----> endpoint PJSIP/bar -- ARI, through the applications resource, can: - subscribe to endpoint:PJSIP/foo and get notifications for channels PJSIP/foo-1,PJSIP/foo-2 and endpoint PJSIP/foo - subscribe to endpoint:PJSIP/bar and get notifications for channels PJSIP/bar-1 and endpoint PJSIP/bar - subscribe to endpoint:PJSIP and get notifications for channels PJSIP/foo-1,PJSIP/foo-2,PJSIP/bar-1 and endpoints PJSIP/foo,PJSIP/bar Note that since endpoint PJSIP never changes, it never has events itself. It merely provides an aggregation point for all other endpoints in its technology (which in turn aggregate all channel messages associated with that endpoint). This patch also adds endpoints to res_xmpp and chan_motif, because the actual messaging work will need it (messaging without XMPP is just sad). Review: https://reviewboard.asterisk.org/r/3760/ ASTERISK-23692 ........ Merged revisions 419196 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419203 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	a2c912e997	media formats: re-architect handling of media for performance improvements ... In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	b299052e20	ari: Add a copy operation for stored recordings ... This patch adds a new operation for stored recordings, copy. It takes an existing stored recording and makes a copy of it in the same directory or a relative directory under the stored recording directory. /ari/recordings/stored/{recordingName}/copy?destinationRecordingName={copy_name} This is particularly useful for voicemail-esque applications, which may need to copy or move recordings around a directory structure. Review: https://reviewboard.asterisk.org/r/3768/ ASTERISK-24036 #close Reported by: Sam Galarneau Tested by: Sam Galarneau ........ Merged revisions 419021 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	d4b436d0ea	ARI/res_stasis: Subscribe to both Local channel halves when originating to app ... This patch fixes two bugs: 1. When originating a channel into a Stasis application, we already create a subscription for the channel that is going into our Stasis app. Unfortunately, when you create a Local channel and pass it off to a Stasis app, you really aren't creating just one channel: you're creating two. This patch snags the second half of the Local channel pair (assuming it is a Local channel pair, but luckily core_local is kind about such assumptions) and subscribes to it as well. 2. Subscriptions are a bit sticky right now. If a subscription is made, the 'interest' count gets bumped on the Stasis subscription - but unless something explicitly unsubscribes the channel, said subscription sticks around. This is not much of a problem is a user is creating the subscription - if they made it, they must want it. However, when we are creating implicit subscriptions, we need to make sure something clears them out. This patch takes a pessimistic approach: it watches the cache updates coming from Stasis and, if we notice that the cache just cleared out an object, we delete our subscription object. This keeps our ao2 container of Stasis forwards in an application from growing out of hand; it also is a bit more forgiving for end users who may not realize they were supposed to unsubscribe from that channel that just hung up. Review: https://reviewboard.asterisk.org/r/3710/ #ASTERISK-23939 #close ........ Merged revisions 418089 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	9b10813a2b	res_ari: Fix some off-nominal paths just dropping the HTTP connection. ... * Removed some incorrect newlines on ast_http_error() messages in manager.c. * Removed an incorrect newline in res_ari_channels.c. Addendum to ASTERISK-23552 ........ Merged revisions 417932 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Sam Galarneau	aa370d6105	ARI: Improvements to body parameters documentation ... The variables body parameter under the originate and originate with id operations of the channel resource showed invalid JSON in its description. The variables body parameter under the userEvent operation of the event resource made no mention that the custom key/value pairs should be wrapped in a variables key in order to be added to the custom user event. ASTERISK-23975 #close Review: https://reviewboard.asterisk.org/r/3692/ ........ Merged revisions 417878 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	a69e0cd32a	ARI: Remove unnecessary \briefs from automatically generated documentation ... Review: https://reviewboard.asterisk.org/r/3440/ ........ Merged revisions 412653 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	365ae7523b	res_http_websocket: Close websocket correctly and use careful fwrite ... When a client takes a long time to process information received from Asterisk, a write operation using fwrite may fail to write all information. This causes the underlying file stream to be in an unknown state, such that the socket must be disconnected. Unfortunately, there are two problems with this in Asterisk's existing websocket code: 1. Periodically, during the read loop, Asterisk must write to the connected websocket to respond to pings. As such, Asterisk maintains a reference to the session during the loop. When ast_http_websocket_write fails, it may cause the session to decrement its ref count, but this in and of itself does not break the read loop. The read loop's write, on the other hand, does not break the loop if it fails. This causes the socket to get in a 'stuck' state, preventing the client from reconnecting to the server. 2. More importantly, however, is that the fwrite in ast_http_websocket_write fails with a large volume of data when the client takes awhile to process the information. When it does fail, it fails writing only a portion of the bytes. With some debugging, it was shown that this was failing in a similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite with a long enough timeout solved the problem. Note that this version of the patch, unlike r417310 in Asterisk 11, exposes configuration options beyond just chan_sip's sip.conf. Configuration options to configure the write timeout have also been added to pjsip.conf and ari.conf. #ASTERISK-23917 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3624/ ........ Merged revisions 417310 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 417311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9cc1a8e893	stasis: Reduce creation of channel snapshots to improve performance ... During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	53968c00b3	TALK_DETECT: A channel function that raises events when talking is detected ... This patch adds a new channel function TALK_DETECT that, when set on a channel, causes events indicating the start/stop of talking on a channel to be emitted to both AMI and ARI clients. The function allows setting both the silence threshold (the length of silence after which we decide no one is talking) as well as the talking threshold (the amount of energy that counts as talking). Parameters can be updated on a channel after talk detection has been enabled, and talk detection can be removed at any time. The events raised by the function use a nomenclature similar to existing AMI/ARI events. For AMI: ChannelTalkingStart/ChannelTalkingStop For ARI: ChannelTalkingStarted/ChannelTalkingFinished Review: https://reviewboard.asterisk.org/r/3563/ #ASTERISK-23786 #close Reported by: Matt Jordan ........ Merged revisions 414934 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414935 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	cf21644d6a	ARI: Add ability to raise arbitrary User Events ... User events can now be generated from ARI. Events can be signalled with arbitrary json variables, and include one or more of channel, bridge, or endpoint snapshots. An application must be specified which will receive the event message (other applications can subscribe to it). The message will also be delivered via AMI provided a channel is attached. Dialplan generated user event messages are still transmitted via the channel, and will only be received by a stasis application they are attached to or if the channel is subscribed to. This change also introduces the multi object blob mechanism used to send multiple snapshot types in a single message. The dialplan app UserEvent was also changed to use multi object blob, and a new stasis message type created to handle them. ASTERISK-22697 #close Review: https://reviewboard.asterisk.org/r/3494/ ........ Merged revisions 414405 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	b9d7dfcc62	ARI: Make bridges/{bridgeID}/play queue sound files ... Previously multiple play actions against a bridge at one time would cause the sounds to play simultaneously on the bridge. Now if a sound is already playing, the play action will queue playback to occur after the completion of other sounds currently on the queue. (closes issue ASTERISK-22677) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/3379/ ........ Merged revisions 412639 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	9a85fc0aa0	ARI: Add debug logging for events and responses ... This adds DEBUG level logging for ARI websocket events and HTTP responses similar to what is available for AMI. Logging for ARI HTTP requests is already adequate for debugging purposes. ........ Merged revisions 412565 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	a8742e327f	ARI: Add tones playback resource ... Adds a tones URI type to the playback resource. The tone can be specified by name (from indications.conf) or by a tone pattern. In addition, tonezone can be specified in the URI (by appending ;tonezone=<zone>). Tones must be stopped manually in order for a stasis control to move on from playback of the tone. Tones may be paused, resumed, restarted, and stopped. They may not be rewound or fast forwarded (tones can't be controlled in a way that lets you skip around from note to note and pausing and resuming will also restart the tone from the beginning). Tests are currently in development for this feature (https://reviewboard.asterisk.org/r/3428/). (closes issue ASTERISK-23433) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3427/ ........ Merged revisions 412535 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	45ade68cb4	Remove unused RAII_VAR() declarations. ... * Remove unused RAII_VAR() declarations. The compiler cannot catch these because the cleanup function "references" the unused variable. Some actually allocated and released resources that were never used. * Fixed some whitespace issues in stasis_bridges.c. ........ Merged revisions 412399 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	d6e2c50058	bridging: Ensure locking during snapshot creation ... While the vast majority of bridge snapshot creation is locked properly, there are currently some instances that are not. This adds the missing locking to ensure bridge state is not malleable during snapshot creation. (closes issue ASTERISK-22904) Review: https://reviewboard.asterisk.org/r/3415/ Reported by: Matt Jordan ........ Merged revisions 412193 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Sean Bright	c32fe8b8e5	ARI: Don't complain about missing ARI users when we aren't enabled ... Currently, if ARI is not enabled it will still complain that there are no configured users. This patch checks to see if ARI is enabled before logging and error or iterating the container to validate the users. Review: https://reviewboard.asterisk.org/r/3391/ ........ Merged revisions 411173 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411174 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	1ba13718fc	assigned-uniqueids: Miscellaneous cleanup and fixes. ... * Fix memory leak in ast_unreal_new_channels(). Made it generate the ;2 uniqueid on a stack variable instead of mallocing it. * Made send error response to ARI and AMI requests instead of just logging excessive uniqueid length and allowing truncation. action_originate() and ari_channels_handle_originate_with_id(). * Fixed minor truncating uniqueid hole when generating the ;2 uniqueid string length. Created public and internal lengths of uniqueid. The internal length can handle a max public uniqueid plus an appended ;2. * free() and ast_free() are NULL tolerant so they don't need a NULL test before calling. * Made use better struct initialization format instead of the position dependent initialization format. Also anything not explicitly initialized in the struct is initialized to zero by the compiler. * Made ast_channel_internal_set_fake_ids() use the safer ast_copy_string() instead of strncpy(). Review: https://reviewboard.asterisk.org/r/3371/ ........ Merged revisions 410949 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	1cf74b8776	res_stasis: Extend bridge type to be a comma separated list of bridge attributes. ... This change turns the bridge type field into a comma separated list of attributes. These attributes include: mixing, holding, dtmf_events, and proxy_media. By setting the various attributes a user can control the type of bridge created with the behavior they need for their application. (closes issue ASTERISK-23437) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3359/ ........ Merged revisions 410904 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	4c2b1c225b	ARI/bridges: Forward Playback/Recording Started/Finished to bridge topic ... (closes issue ASTERISK-23444) Reported by: Ben Merrills Review: https://reviewboard.asterisk.org/r/3340/ ........ Merged revisions 410558 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	ef69b5176d	unqiueid: correct max uniqueid length test ... This patch adds null string test prior to checking for a max uniqueid value that was added in r410157. ........ Merged revisions 410368 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	5ca081e053	resource_channels: Check if a passed in ID is NULL before checking its length ... Calling strlen on a NULL string is explosive. This patch checks whether or not the passed in string is NULL or zero length before checking to see if the string is too long. ........ Merged revisions 410274 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410275 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	80ef9a21b9	uniqueid: channel linkedid, ami, ari object creation with id's ... Much needed was a way to assign id to objects on creation, and much change was necessary to accomplish it. Channel uniqueids and linkedids are split into separate string and creation time components without breaking linkedid propgation. This allowed the uniqueid to be specified by the user interface - and those values are now carried through to channel creation, adding the assignedids value to every function in the chain including the channel drivers. For local channels, the second channel can be specified or left to default to a ;2 suffix of first. In ARI, bridge, playback, and snoop objects can also be created with a specified uniqueid. Along the way, the args order to allocating channels was fixed in chan_mgcp and chan_gtalk, and linkedid is no longer lost as masquerade occurs. (closes issue ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3191/ ........ Merged revisions 410157 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	3f730662f7	res_stasis_recording: Add a "target_uri" field to recording events. ... This change adds a target_uri field to the live recording object. It contains the URI of what is being recorded. (closes issue ASTERISK-23258) Reported by: Ben Merrills Review: https://reviewboard.asterisk.org/r/3299/ ........ Merged revisions 410025 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d277f3ec3e	json: Fix off-nominal json ref counting issues. ... * Fixed off-nominal json ref counting issue with using the following API calls: ast_json_object_set() and ast_json_array_append(). * Fixed off-nominal error reporting in ast_ari_endpoints_list(). * Fixed some miscellaneous off-nominal json ref counting issues in report_receive_fax_status() and dial_to_json(). ........ Merged revisions 408713 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	0ebbeac69f	ari/resource_channels: Add channel variables earlier in the creation process ... This patch tweaks the behaviour of POST /channels with channel variables such that the variables are passed into the pbx.c routines that perform the origination. This allows the variables to be assigned to the newly created channels immediately upon their construction, as opposed to be assigned after the originate has completed. The upshot of this is that the variables are available on the channels if they execute in the dialplan, as opposed to only being available once the channels are answered. Review: https://reviewboard.asterisk.org/r/3183/ ........ Merged revisions 407937 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	e5899852cc	res_stasis: Enable transfers and provide events when they occur. ... This change enables transfers within ARI created bridges and adds events for when they occur. Unlike other events these will be received if *any* subscribed object is involved in the transfer. (closes issue ASTERISK-22984) Reported by: David M. Lee Review: https://reviewboard.asterisk.org/r/3120/ ........ Merged revisions 407153 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	1590d32ab0	ARI: Support channel variables in originate ... This adds back in support for specifying channel variables during an originate without compromising the ability to specify query parameters in the JSON body. This was accomplished by generating the body-parsing code in a separate function instead of being integrated with the URI query parameter parsing code such that it could be called by paths with body parameters. This is transparent to the user of the API and prevents manual duplication of code or data structures. (closes issue ASTERISK-23051) Review: https://reviewboard.asterisk.org/r/3122/ Reported by: Matt Jordan ........ Merged revisions 406003 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	aa9db707c5	ARI: Add mailboxes resource for controlling and polling external MWI ... Adds the following AMI commands: PUT mailboxes/mailboxName modifies mailbox state and implicitly creates new mailboxes GET mailboxes/mailboxName retrieves a JSON representation of a single mailbox if it exists GET mailboxes retrieves a JSON array of all mailboxes DELETE mailbox/mailboxName deletes a mailbox Note that res_mwi_external must be loaded for these functions to actually do anything. Review: https://reviewboard.asterisk.org/r/3117/ ........ Merged revisions 405553 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	8585340b87	res_ari: Fix various memory leaks. ... This change fixes a few memory leaks that were found based on a mailing list post. 1. Some JSON response messages were never freed. This was caused by the documentation stating that message references were stolen when in reality they were not. The code now follows the documentation and usage has been updated. 2. HTTP response headers were never freed. 3. The variable list for wildcards paths was never freed. (closes issue ASTERISK-23128) Reported by: Kenneth Watson (on list) Review: https://reviewboard.asterisk.org/r/3119/ ........ Merged revisions 405325 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405326 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	40a7f68e4b	ari: Remove support for specifying channel vars during origination. ... When we added support for specifying channel variables for an origination, we didn't consider how that would interact with another feature, namely specifying request parameters in a JSON request body. The method of specifying channel variables (as a flat JSON object passed in the JSON body) interferes with parsing parameters out of the request body. Unfortunately, fixing this would be a backward incompatible change. In the interest of keeping the API sane and keeping our release schedule, we're dropping the feature for specifying channel variables in the origination request. We will bring the feature back soon, as a backward compatible addition to the API. (closes issue ASTERISK-23051) Review: https://reviewboard.asterisk.org/r/3088 ........ Merged revisions 404509 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404512 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	7ddfe12aff	res_stasis_device_state: Set resource type for subscriptions to deviceState ... The documentation for ARI already specifies that the device state resource when used for subscribing for events is "deviceState", not "device_state". The code, however, used "device_state"; although this was inconsistent as well in doxygen comments in resource_applications. Because the actual resource being subscribed to is /deviceStates/{device}/, it makes sense for the resource type specifier to be deviceState. Note that the key value in the events is still "device_state". ........ Merged revisions 404437 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404438 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	72c282cc66	ao2_iterator: Mini-audit of the ao2_iterator loops in the new code files. ... * Fixed several places where ao2_iterator_destroy() was not called. * Fixed several iterator loop object variable reference problems. * Fixed res_parking AMI actions returning non-zero. Only the AMI logoff action can return non-zero. Review: https://reviewboard.asterisk.org/r/3087/ ........ Merged revisions 404434 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	b0bb03e916	bridging: Give bridges a name and a known creator ... Bridges have two new optional properties, a creator and a name. Certain consumers of bridges will automatically provide bridges that they create with these properties. Examples include app_bridgewait, res_parking, app_confbridge, and app_agent_pool. In addition, a name may now be provided as an argument to the POST function for creating new bridges via ARI. (closes issue AFS-47) Review: https://reviewboard.asterisk.org/r/3070/ ........ Merged revisions 404042 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	76042b1eb4	ari/resource_channels: When creating a channel, specify a default format (SLIN) ... When creating channels via ARI, the current code fails to provide any default format capabilities. For non-virtual channels this isn't really a problem - the channels typically receive their capabilities as a result of the underlying channel driver configuration. For virtual channels (such as Local channels), the lack of any format capabilities causes the Asterisk core to make some 'odd' choices with respect to the translation paths. The issue reporter had some paths that had 3 hops on each channel leg, causing multiple transcodings and some really crappy audio/performance. By specifying a baseline of SLIN, we prevent that from occurring. Note that this is what AMI does when it performs an Originate, as does res_clioriginate. Review: https://reviewboard.asterisk.org/r/3068/ (issue ASTERISK-22962) Reported by: Matt DiMeo ........ Merged revisions 403993 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403994 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	3a5cc054ed	res_stasis: Expose event for call forwarding and follow forwarded channel. ... This change adds an event for when an originated call is redirected to another target. This event contains the original channel and the newly created channel. If a stasis subscription exists on the original originated channel for a stasis application then a new subscription will also be created on the stasis application to the redirected channel. This allows the application to follow the call path completely. (closes issue ASTERISK-22719) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/3054/ ........ Merged revisions 403808 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago