asterisk

Commit Graph

Author	SHA1	Message	Date
Jonathan Rose	a6329a3acf	ARI: MOH start and stop for a channel (issue ASTERISK-21974) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2680/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	17c546173f	ARI: Bridge Playback, Bridge Record Adds a new channel driver for creating channels for specific purposes in bridges, primarily to act as either recorders or announcers. Adds ARI commands for playing announcements to ever participant in a bridge as well as for recording a bridge. This patch also includes some documentation/reponse fixes to related ARI models such as playback controls. (closes issue ASTERISK-21592) Reported by: Matt Jordan (closes issue ASTERISK-21593) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2670/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	5a8f32703c	Filter channels used as internal mechanisms This adds new flags to the channel tech properties that flag it as different types of implementation detail used exclusively to provide a feature. Examples of channels that would have these flags include the announcement and recording channels used by confbridge which are the only two marked as such by this patch. Review: https://reviewboard.asterisk.org/r/2633/ (closes issue ASTERISK-21873) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	c47787feab	Add a bunch of options from sip.conf to res_sip.conf For a complete list of the options added, see the review linked at the bottom of this commit message. (closes issue ASTERISK-21506) reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2671 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	3c86832f9f	Fixed null dereference when WebSocket subprotocol isn't specified git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	c1a7567d24	ARI: Add support for suppressing media streams. Also convert res_mutestream to use the core feature behind this. (closes issue ASTERISK-21618) Review: https://reviewboard.asterisk.org/r/2652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	6d9909887e	Properly indicate failure to open an audio stream in res_agi If there is an error streaming an audio file, the current return status makes it difficult for an AGI script to determine that there was an error with the audio file. This patches changes the result to return -1 and the function returns RESULT_FAILURE instead of RESULT_SUCCESS. From looking at other parts of res_agi, this would appear to be the proper way to handle an error. (closes issue ASTERISK-21903) Reported by: Ariel Wainer Tested by: Ariel Wainer Patches: asterisk-21903-return-stream-res_1.8.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2625/ ........ Merged revisions 394640 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 394641 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394642 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	e1b75afdb8	Debug logging to help with WebSocket connection problems git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394513 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	6bdd453168	Prevent crash from trying to end a session in an invalid way. This ensures that code that was only meant to be run on a reinvite failure only runs on a reinvite failure. (closes issue ASTERISK-22061) reported by Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	80dd0229f1	Fixed null dereference when WebSocket protocol is omitted git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394442 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	de31a362db	Document the ari.conf allowed_origins setting git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394397 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	b75b88e8f7	Remove some callbacks and functions which are not needed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	2bad69006f	ARI: Add support for Cross-Origin Resource Sharing (CORS), origin headers This rejects requests from any unknown origins. (closes issue ASTERISK-21278) Review: https://reviewboard.asterisk.org/r/2667/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394189 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	0ce29906eb	Tweak the subscription failure warning message to include endpoint name and context. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394103 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	684481b74c	Change ARI user config to use a type field When I initially wrote the configuration support for ARI users, I determined the section type by a category prefix (i.e., [user-admin]). This is neither idiomatic Asterisk configuration, nor is it really that user friendly. This patch replaces the category prefix with a type field in the section, which is much cleaner. Review: https://reviewboard.asterisk.org/r/2664/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394076 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	fb09d5bc60	Apply defaults to ari.conf's general section git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394065 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	c232a1f821	Handle outbound registration failures that do not occur as a result of a real response. (closes issue ASTERISK-22064) Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394004 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	15036c2979	Document the 400 error response for originate git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	a73394abb8	Corrected api-docs for channel variables git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393968 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	6e192a0f6f	Ensure all pjsip_regc_* access occurs within a pjlib thread. (closes issue ASTERISK-22054) Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	6670f56027	Tweak log message slightly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	0f7d4d308e	Treat the authentication object as invalid if digest configuration is chosen and the digest is not of the correct length. (closes issue ASTERISK-22003) Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393857 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	a0684d97f5	Oh menuconfig, why do you hate margins? git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393843 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	ed60f4793a	Better structure for the WebSocket validation failure message git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	259fb38187	Ensure that a valid bind host is specified for transports. (closes issue ASTERISK-22017) Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	7c044acbd9	Refactor operations to access the stasis cache instead of objects directly when retrieving information. (closes issue ASTERISK-21883) Review: https://reviewboard.asterisk.org/r/2645/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	b698d80d4b	res_stasis_http doesn't depend on res_stasis any more git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	b083a4cdae	res_parking: Apply ringing role option on swap with a channel that rings (closes issue ASTERISK-21877) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2656/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393815 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	7ee5b025f4	Fix building. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	87973eecff	ARI: Add support for getting/setting channel and global variables. This allows for reading and writing of functions on channels. (closes issue ASTERISK-21868) Review: https://reviewboard.asterisk.org/r/2641/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393806 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	7422581b6d	Move channel driver Registry manager events to core. This also shuffles the stasis system topic and related handling. (closes issue ASTERISK-21488) Review: https://reviewboard.asterisk.org/r/2631/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	7fdeb52910	Fix some broken logic in sending outbound caller ID. * trust_id_outbound was required even when the caller ID was not marked private. This is against intentions and documentation. * We now check both name and number privacy instead of checking name privacy twice. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	c54b26a18c	ARI: return a 503 if Asterisk isn't fully booted git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	cc3478d2e8	Print error details when set nonblock fails git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	01c21c7aea	Document MissingParams error message for /ari/events git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	d0a55fa52d	Refactor RTCP events over to Stasis; associate with channels This patch does the following: * It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel information in the RTCP events. Because Stasis provides a cache, Jaco's patch was modified to pass the channel uniqueid to the RTP layer as opposed to a pointer to the channel. This has the following benefits: (1) It keeps the RTP engine 'clean' of references back to channels (2) It prevents circular dependencies and other potential ref counting issues * The RTP engine now allows any RTP implementation to raise RTCP messages. Potentially, other implementations (such as res_rtp_multicast) could also raise RTCP information. The engine provides structs to represent RTCP headers and RTCP SR/RR reports. * Some general refactoring in res_rtp_asterisk was done to try and tame the RTCP code. It isn't perfect - that's way beyond the scope of this work - but it does feel marginally better. * A few random bugs were fixed in the RTCP statistics. (Example: performing an assignment of a = a is probably not correct) * We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't raise an event when we sent a RR report. Note that this work will be of use to others who want to monitor call quality or build modules that report call quality statistics. Since the events are now moving across the Stasis message bus, this is far easier to accomplish. It is also a first step (though by no means the last step) towards getting Olle's pinefrog work incorporated. Again: note that the patch by Jaco Kroon was modified slightly for this work; however, he did all of the hard work in finding the right places to set the channel in the RTP engine across the channel drivers. Much thanks goes to Jaco for his hard work here. Review: https://reviewboard.asterisk.org/r/2603/ (closes issue ASTERISK-20574) Reported by: Jaco Kroon patches: asterisk-rtcp-channel.patch uploaded by jkroon (License 5671) (closes issue ASTERISK-21471) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	93ed5ef0ff	res_parking: Replace Parker snapshots with ParkerDialString This process also involved a large amount of rework regarding how to redial the Parker when a channel leaves a parking lot due to timeout. An attended transfer channel variable has been added to attended transfers to extensions that will eventually park (but haven't at the time of transfer) as well. This resolves one of the two BUGBUG comments remaining in res_parking. (issues ASTERISK-21877) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2638/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393704 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	fb03bf9b39	Fix int width problem for 32-bit... again git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	dbc588b02f	Fix load errors related to the new ari_model_validators. The Asterisk strategy of loading modules with RTLD_LAZY to extract metadata from the module works well enough, until you try to take the address of a function. If a module takes the address of a function, that function needs to be resolved at load time. That kinda defeats RTLD_LAZY. This patch adds some ari_validator_{id}_fn() wrapper functions for safely getting the function pointer from a different module. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	ef032842f1	Violating the margins to make menuconfig happy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393561 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	a75fd32212	ARI - channel recording support This patch is the first step in adding recording support to the Asterisk REST Interface. Recordings are stored in /var/spool/recording. Since recordings may be destructive (overwriting existing files), the API rejects attempts to escape the recording directory (avoiding issues if someone attempts to record to ../../lib/sounds/greeting, for example). (closes issue ASTERISK-21594) (closes issue ASTERISK-21581) Review: https://reviewboard.asterisk.org/r/2612/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	9ba976b19c	ARI authentication. This patch adds authentication support to ARI. Two authentication methods are supported. The first is HTTP Basic authentication, as specified in RFC 2617[1]. The second is by simply passing the username and password as an ?api_key query parameter (which allows swagger-ui[2] to authenticate more easily). ARI usernames and passwords are configured in the ari.conf file (formerly known as stasis_http.conf). The user may be set to `read_only`, which will prohibit the user from issuing POST, DELETE, etc. Also, the user's password may be specified in either plaintext, or encrypted using the crypt() function. Several other notes about the patch. * A few command line commands for seeing ARI config and status were also added. * The configuration parsing grew big enough that I extracted it to its own file. [1]: http://www.ietf.org/rfc/rfc2617.txt [2]: https://github.com/wordnik/swagger-ui (closes issue ASTERISK-21277) Review: https://reviewboard.asterisk.org/r/2649/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	c9a3d4562d	Update events to use Swagger 1.3 subtyping, and related aftermath This patch started with the simple idea of changing the /events data model to be more sane. The original model would send out events like: { "stasis_start": { "args": [], "channel": { ... } } } The event discriminator was the field name instead of being a value in the object, due to limitations in how Swagger 1.1 could model objects. While technically sufficient in communicating event information, it was really difficult to deal with in terms of client side JSON handling. This patch takes advantage of a proposed extension[1] to Swagger which allows type variance through the use of a discriminator field. This had a domino effect that made this a surprisingly large patch. [1]: https://groups.google.com/d/msg/wordnik-api/EC3rGajE0os/ey_5dBI_jWcJ In changing the models, I also had to change the swagger_model.py processor so it can handle the type discriminator and subtyping. I took that a big step forward, and using that information to generate an ari_model module, which can validate a JSON object against the Swagger model. The REST and WebSocket generators were changed to take advantage of the validators. If compiled with AST_DEVMODE enabled, JSON objects that don't match their corresponding models will not be sent out. For REST API calls, a 500 Internal Server response is sent. For WebSockets, the invalid JSON message is replaced with an error message. Since this took over about half of the job of the existing JSON generators, and the .to_json virtual function on messages took over the other half, I reluctantly removed the generators. The validators turned up all sorts of errors and inconsistencies in our data models, and the code. These were cleaned up, with checks in the code generator avoid some of the consistency problems in the future. * The model for a channel snapshot was trimmed down to match the information sent via AMI. Many of the field being sent were not useful in the general case. * The model for a bridge snapshot was updated to be more consistent with the other ARI models. Another impact of introducing subtyping was that the swagger-codegen documentation generator was insufficient (at least until it catches up with Swagger 1.2). I wanted it to be easier to generate docs for the API anyways, so I ported the wiki pages to use the Asterisk Swagger generator. In the process, I was able to clean up many of the model links, which would occasionally give inconsistent results on the wiki. I also added error responses to the wiki docs, making the wiki documentation more complete. Finally, since Stasis-HTTP will now be named Asterisk REST Interface (ARI), any new functions and files I created carry the ari_ prefix. I changed a few stasis_http references to ari where it was non-intrusive and made sense. (closes issue ASTERISK-21885) Review: https://reviewboard.asterisk.org/r/2639/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	dcf03554a0	Shuffle RESTful URL's around. This patch moves the RESTful URL's around to more appropriate locations for release. The /stasis URL's are moved to /ari, since Asterisk REST Interface was a more appropriate name than Stasis-HTTP. (Most of the code still has stasis_http references, but they will be cleaned up after there are no more outstanding branches that would have merge conflicts with such a change). A larger change was moving the ARI events WebSocket off of the shared /ws URL to its permanent home on /ari/events. The Swagger code generator was extended to handle "upgrade: websocket" and "websocketProtocol:" attributes on an operation. The WebSocket module was modified to better handle WebSocket servers that have a single registered protocol handler. If a client connections does not specify the Sec-WebSocket-Protocol header, and the server has a single protocol handler registered, the WebSocket server will go ahead and accept the client for that subprotocol. (closes issue ASTERISK-21857) Review: https://reviewboard.asterisk.org/r/2621/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	d98c19f947	Add pjproject dependency to res_sip_notify git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	a25a630659	New SIP Channel driver: Always Auth Reject If no matching endpoint is found for the incoming request Asterisk will respond with a 401 Unauthorized (rejecting the request), but will first challenge if no authorization creditials are given. Changes also included moving ACL options into a new global 'security' configuration section in res_sip.conf. (closes issue ASTERISK-21433) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2554/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393442 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	5456794b66	New SIP Channel Driver - Add CLI/AMI initiated NOTIFY requests Added the ability to send unsolicited NOTIFY requests to a particular endpoint with a configured payload. Added both CLI and AMI support. For a given endpoint, this module will iterate over all its contacts sending the appropriate NOTIFY request to each. (closes issue ASTERISK-21436) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2623/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	f820d24db1	ARI: Implement channel hold/unhold. This puts the channel on hold (rather than queueing a frame from the channel). (closes issue ASTERISK-21619) Review: https://reviewboard.asterisk.org/r/2647/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	f41faf0b7d	ARI: Implement channel dial. This creates a new outbound channel, and bridges it to a channel already in the Stasis application. (closes issue ASTERISK-21620) Review: https://reviewboard.asterisk.org/r/2634/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393326 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	909ee4bfb9	Refactor extraneous channel events This change removes JitterBufStats, ChannelReload, and ChannelUpdate and refactors the following events to travel over Stasis-Core: * LocalBridge * DAHDIChannel * AlarmClear * SpanAlarmClear * Alarm * SpanAlarm * DNDState * MCID * SIPQualifyPeerDone * SessionTimeout Review: https://reviewboard.asterisk.org/r/2627/ (closes issue ASTERISK-21476) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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2082 Commits (bdb1c6bfb03b5734f637f8678c1bc08910c1bc82)