24119 Commits (02f55a36a0b8e38631c90cb3465085fbc6ed796c)

 

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Author	SHA1	Message	Date
Richard Mudgett	02f55a36a0	Revert accidental overcommit. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393632 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	b4e9a3fc2f	Add BUGBUG note for ASTERISK-22009 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393631 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	c841c34aae	chan_dahdi: Fix segfault reloading chan_dahdi when round robin is used. ... * Clear round_robin[] in dahdi_restart(). (closes issue ASTERISK-21847) Reported by: Ivo Andonov Patches: jira_asterisk_21847_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 393627 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 393628 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393629 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	0227e00eb3	OneTouchRecord: Make so Monitor/MixMonitor can be toggled/started/stopped. ... The OneTouchRecord feature has historically been a toggle. This patch adds the ability to make the OneTouchRecord hook optionally start/stop recording only. If OneTouchRecord is already doing what is requested then only the invoker hears the courtesy tone and/or start/stop recording message. The new feature is written so we could easily add explicit start/stop recording DTMF hooks for Monitor and MixMonitor. The majority of the changes in bridge_builtin_features.c is a refactoring of the OneTouchRecord code (Monitor and MixMonitor versions) so it is easy to direct the toggle/start/stop functionality. Review: https://reviewboard.asterisk.org/r/2655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	ad5dc3c159	Move when bridge channel enter is published so it does not interrupt the thought of some lines of code. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393601 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	b96d8cbc78	Fix some indentation in stasis_config.c. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	a981de9adb	Fix some bugs in CDRs; add some CLI commands to help debugging ... This patch fixes a few minor bugs and one major one: the CDR by bridge container was less than helpful. The mechanism previously used to try and find all of the CDRs in a particular bridge ended up missing CDRs, resulting in incorrect records. When looking up CDRs in a bridge, we now just bite the bullet and do a selection across all existing CDRs. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393599 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	569f5f2117	Let Stasis load itself with default values ... While a Stasis configuration file is nice, it shouldn't be mandatory. We can carry on with default values. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	0f725bd5d9	Publish a bridge enter before pulling on a push-and-swap operation. ... Prior to this patch, the order of procedures on a bridge push was * Add new bridge channel to bridge's array. * Pull the swap channel out of the bridge * Publish a bridge enter event. The problem is that when the swap channel was pulled from the bridge, a bridge leave event would be published. The bridge snapshot published during the bridge leave showed the new channel that had been added to the bridge, but there had been no bridge enter event for that channel. The fix provided here was to change the order a bit * Add new bridge channel to bridge's array. * Publish bridge enter event. * Pull the swap channel out of the bridge. This makes it so that the bridge snapshots during the stasis events are accurate. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 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	12 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	12 years ago
David M. Lee	c4adaf9106	Configuration for Stasis threadpool ... The appropriate settings for the Stasis threadpool is very system specific, depending upon both workload and system configuration. This patch adds a stasis.conf file which can be used to configure the key attributes of the threadpool for the Stasis message bus. (closes issue ASTERISK-21280) Review: https://reviewboard.asterisk.org/r/2651/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 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	12 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	12 years ago
Jason Parker	85ba063329	Add a SystemName field to all AMI events. ... This only gets sent out if configured in asterisk.conf (closes issue ASTERISK-21494) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	a3c91e955a	MixMonitor: Minor code cleanup. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393500 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	e4a5ae0376	MixMonitor: Make start_mixmonitor_callback() options parameter NULL tolerant. ... * Removed some unnecessary code in start_mixmonitor_callback(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393496 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	f6cdd9a12c	MixMonitor: Don't use ast_strdupa() in a loop. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	b47152121c	MixMonitor: Update XML documentation and CLI "mixmonitor {start|stop|list}" help. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	8ab7724352	MixMonitor: Fix refleak in manager_stop_mixmonitor() if could not stop monitoring. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	6a65c4d072	MixMonitor: Remove some unnecessary channel locking. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	631fad018f	Fix MixMonitor b option. ... The option had not been converted to use the replacement for ast_bridged_channel(). One touch mixmonitor now records files again. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	96fed373e9	Fix chan_gtalk.c compile error. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 years ago
Mark Michelson	69242ad317	Remove unused blind transfer publication structure. ... I ended up using a bridge blob, so this structure was unused. Keeping it in the header would just cause confusion. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393463 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	05a16729cb	Stasis - Refactor AOC Events ... Refactored the AMI events in AOC onto Stasis-Core. The ast_aoc_manager_event function now publishes a channel snapshot, along with a JSON blob describing the advice of charge. A "to_ami" handler has also been added that converts the channel snapshot and AOC event data back into the appropriate data structure for use with AMI. (closes issue ASTERISK-21472) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2643/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 years ago
Kinsey Moore	de206baa99	Fix transfer AMI event parameter naming ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393429 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	6e3a5d2d48	Add CEL unit tests and do some cleanup ... This adds several unit tests for CEL functionality and provides the requisite framework for creating additional unit tests. This also cleans up some reference leaks that were occurring in Stasis-Core message callback code. Review: https://reviewboard.asterisk.org/r/2646/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393410 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	f7624718f8	Fix issue with inability to cancell call transfer made by on-sceen menus. ...	12 years ago
Tzafrir Cohen	51c54ddf8d	ast_tls_cert: don't recreate generated files ... Don't regenrate cat.cfg, ca.crt and ca.key if they were already created on a previous run. (closes issue ASTERISK-21932) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 years ago
Matthew Jordan	3841520a6e	Prevent crash during synchronous AMI origination by ref bumping returned channel ... The originate APIs allow callers to provide a pointer to a channel that will point to the originated channel if the function call succeeds. This is used by AMI to provide channel information when the originate is performed synchronously. Unfortunately, if the originate fails in certain ways, the outbound channel is already disposed of during the dialing itself. This results in the channel being improperly dereferenced by the internal originate function in pbx.c. This patch ref bumps the channel to prevent this from occurring. Callers must now unlock and unref the channel (which is more in line with general channel management guidelines anyway). This only affects manager, as it is the only consumer of this API function that actually passes in a channel pointer. Review: https://reviewboard.asterisk.org/r/2617/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393361 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 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	12 years ago
Jonathan Rose	f306dbd841	bridge_features: Support One touch Monitor/MixMonitor ... In addition to porting those features, they now enjoy greater feature parity with one another. Specifically, AutoMixMon now has a start and stop message that can be specified with TOUCH_MIXMONITOR_MESSAGE_START and TOUCH_MIXMONITOR_MESSAGE_STOP. (closes issue ASTERISK-21553) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2620/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 years ago
Joshua Colp	68b3bce8b8	Nothing to see here, move along. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393264 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	f523a96635	Implement the defined PUBLISH ESC API within res_sip_pubsub. ... (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	12 years ago
Richard Mudgett	a174aa73f6	Tweak after bridge callback reason to string strings. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	812abf0554	Fix after bridge callback datastore data memory leak. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	e89e95228f	This is no longer needed. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393239 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	e72b009928	Promote local channel optimizing debug messages to verbose 3 messages. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	50ff1f5fc1	res_parking: Dynamic Parking Lots ... (closes issue ASTERISK-21644) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2615/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	84395ff042	features: call pickup stasis refactoring ... (issue ASTERISK-21544) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2588/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3aa6e93d8f	Fix overlapping enum ast_bridge_feature_flags. ... Things may no longer behave in an unexpected fashion. Local channel optimization to holding bridges will work again. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393184 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	6d624eb008	Add stasis publications for blind and attended transfers. ... This creates stasis messages that are sent during a blind or attended transfer. The stasis messages also are converted to AMI events. Review: https://reviewboard.asterisk.org/r/2619 (closes issue ASTERISK-21337) Reported by Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	ca61a05506	Handle an originated channel being sent into a non-empty bridge ... Originated channels are a bit odd - they are technically a dialed channel (thus the party B or peer) but, since there is no caller, they are treated as the party A. When entering into a bridge that already contains participants, the CDR engine - if the CDR record is in the Dial state - attempts to match the person entering the bridge with an existing participant. The idea is that if you dialed someone and the person you dialed is already in the bridge, you don't need a new CDR record, the existing CDR record describes the relationship. Unfortunately, for an originated channel, there is no Party B. If no one was in the bridge this didn't cause any issues; however, if participants were in the bridge the CDR engine would attempt to match a non-existant Party B on the channel's CDR record and explode. This patch fixes that, and a unit test has been added to cover this case. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago