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2013-07-19Filter channels used as internal mechanismsKinsey Moore
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
2013-07-18Add a bunch of options from sip.conf to res_sip.confMark Michelson
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
2013-07-18Fixed null dereference when WebSocket subprotocol isn't specifiedDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-18ARI: Add support for suppressing media streams.Jason Parker
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
2013-07-18Properly indicate failure to open an audio stream in res_agiMichael L. Young
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
2013-07-16Debug logging to help with WebSocket connection problemsDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-16Prevent crash from trying to end a session in an invalid way.Mark Michelson
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
2013-07-16Fixed null dereference when WebSocket protocol is omittedDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-15Document the ari.conf allowed_origins settingDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-15Remove some callbacks and functions which are not needed.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-12ARI: Add support for Cross-Origin Resource Sharing (CORS), origin headersJason Parker
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
2013-07-11Tweak the subscription failure warning message to include endpoint name and ↵Joshua Colp
context. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-11Change ARI user config to use a type fieldDavid M. Lee
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
2013-07-11Apply defaults to ari.conf's general sectionDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394065 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-10Handle outbound registration failures that do not occur as a result of a ↵Joshua Colp
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
2013-07-10Document the 400 error response for originateDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-10Corrected api-docs for channel variablesDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-09Ensure all pjsip_regc_* access occurs within a pjlib thread.Joshua Colp
(closes issue ASTERISK-22054) Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-08Tweak log message slightly.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-08Treat the authentication object as invalid if digest configuration is chosen ↵Joshua Colp
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
2013-07-08Oh menuconfig, why do you hate margins?David M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-08Better structure for the WebSocket validation failure messageDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-08Ensure that a valid bind host is specified for transports.Joshua Colp
(closes issue ASTERISK-22017) Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-08Refactor operations to access the stasis cache instead of objects directly ↵Joshua Colp
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
2013-07-08res_stasis_http doesn't depend on res_stasis any moreDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393816 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-08res_parking: Apply ringing role option on swap with a channel that ringsJonathan Rose
(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
2013-07-08Fix building.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-08ARI: Add support for getting/setting channel and global variables.Jason Parker
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
2013-07-08Move channel driver Registry manager events to core.Jason Parker
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
2013-07-08Fix some broken logic in sending outbound caller ID.Mark Michelson
* 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
2013-07-05ARI: return a 503 if Asterisk isn't fully bootedDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-05Print error details when set nonblock failsDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-05Document MissingParams error message for /ari/eventsDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-05Refactor RTCP events over to Stasis; associate with channelsMatthew Jordan
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
2013-07-04res_parking: Replace Parker snapshots with ParkerDialStringJonathan Rose
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
2013-07-04Fix int width problem for 32-bit... againDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-03Fix load errors related to the new ari_model_validators.David M. Lee
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
2013-07-03Violating the margins to make menuconfig happyDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-03ARI - channel recording supportDavid M. Lee
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
2013-07-03ARI authentication.David M. Lee
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
2013-07-03Update events to use Swagger 1.3 subtyping, and related aftermathDavid M. Lee
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
2013-07-03Shuffle RESTful URL's around.David M. Lee
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
2013-07-02Add pjproject dependency to res_sip_notifyDavid M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-02New SIP Channel driver: Always Auth RejectKevin Harwell
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
2013-07-01New SIP Channel Driver - Add CLI/AMI initiated NOTIFY requestsKevin Harwell
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
2013-07-01ARI: Implement channel hold/unhold.Jason Parker
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
2013-07-01ARI: Implement channel dial.Jason Parker
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
2013-07-01Refactor extraneous channel eventsKinsey Moore
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
2013-06-29Nothing to see here, move along.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-29Implement the defined PUBLISH ESC API within res_sip_pubsub.Joshua Colp
(closes issue ASTERISK-21452) Review: https://reviewboard.asterisk.org/r/2630/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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