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2013-07-19Fix crash when using temporary peersKinsey Moore
Temporary peers do not have an associated Stasis endpoint and quite a bit of code in chan_sip assumes that all peers have a Stasis endpoint. All endpoint accesses in chan_sip are now wrapped in an endpoint NULL-check. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394795 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
2013-07-17Left over spacing issues of review 726.Tzafrir Cohen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-17handle DAHDI_EVENT_REMOVED on a pri D-ChannelTzafrir Cohen
When a DAHDI device is removed at run-time it sends the event DAHDI_EVENT_REMOVED on each channel. This is intended to signal the userspace program to close the respective file handle, as the driver of the device will need all of them closed to properly clean-up. This event has long since been handled in chan_dahdi (chan_zap at the time). However the event that is sent on a D-Channel of a "PRI" (ISDN) span simply gets ignored. This commit adds handling for closing the file descriptor (and shutting down the span, while we're at it). It also adds a CLI command 'pri destroy span <N>' to destroy the span and its DAHDI channels. Review: https://reviewboard.asterisk.org/r/726/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-16chan_gulp: Fix gulp_indicate() handling of AST_CONTROL_PVT_CAUSE_CODE.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-16Remove some dead code dealing with old bridging method.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-15Replace chan_agent with app_agent_pool.Richard Mudgett
The ill conceived chan_agent is no more. It is now replaced by app_agent_pool. Agents login using the AgentLogin() application as before. The AgentLogin() application no longer does any authentication. Authentication is now the responsibility of the dialplan. (Besides, the authentication done by chan_agent did not match what the voice prompts asked for.) Sample extensions.conf [login] ; Sample agent 1001 login ; Set COLP for in between calls so the agent does not see the last caller COLP. exten => 1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>) ; Give the agent DTMF transfer and disconnect features when connected to a caller. same => n,Set(CHANNEL(dtmf-features)=TX) same => n,AgentLogin(1001) same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS}) same => n,Hangup() [caller] ; Sample caller direct connect to agent 1001 exten => 800,1,AgentRequest(1001) same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS}) same => n,Hangup() ; Sample caller going through a Queue to agent 1001 exten => 900,1,Queue(agent_q) same => n,Hangup() Sample queues.conf [agent_q] member => Local/800@caller,,SuperAgent,Agent:1001 Under the hood operation overview: 1) Logged in agents wait for callers in an agents holding bridge. 2) Caller requests an agent using AgentRequest() 3) A basic bridge is created, the agent is notified, and caller joins the basic bridge to wait for the agent. 4) The agent is either automatically connected to the caller or must ack the call to connect. 5) The agent is moved from the agents holding bridge to the basic bridge. 6) The agent and caller talk. 7) The connection is ended by either party. 8) The agent goes back to the agents holding bridge. To avoid some locking issues with the agent holding bridge, I needed to make some changes to the after bridge callback support. The after bridge callback is now a list of requested callbacks with the last to be added the only active callback. The after bridge callback for failed callbacks will always happen in the channel thread when the channel leaves the bridging system or is destroyed. (closes issue ASTERISK-21554) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2657/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-13Pretty up a debug message if the referred-by-uri isn't availableMatthew Jordan
Instead of formatting a NULL pointer into a "%s" format string (which is usually not a good thing to do), we instead print "Unknown". git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-12Fix a longstanding issue with MFC-R2 configuration that prevented usersMoises Silva
from mixing different variants or general MFC-R2 settings within the same E1 line. Most users do not have a problem with this since MFC-R2 lines are usually fractional E1s, or the whole E1 has the same country variant and R2 settings. In Venezuela however is common to have inbound MFC-R2 and outbound DTMF-R2 within the same E1. This fix now properly parses the chan_dahdi.conf file to generate a new openr2 context every time a new channel => section is found and the configuration was changed. (closes issue ASTERISK-21117) Reported by: Rafael Angulo Related Elastix issue: http://bugs.elastix.org/view.php?id=1612 ........ Merged revisions 394106 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 394173 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-12Fix a compiler warning.Jason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-11Fixed chan_skinny for systems were pthread_t isn't an int.David M. Lee
I'm looking at you, OS X. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394156 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-11Refactor and cleanup of skinny session handling.Damien Wedhorn
Major changes are to pull all packet reading functions into skinny_session and move timeout handling to scheduling arrangements. Thread cancelling is now undertaken directly rather than waiting for the read to timeout (cleanup is popped on thread cancel). Also added some keepalive timings in debugging messages. Keepalive timeout has been increased from 1.1 by keepalive to 3 times keepalive. This seems to align (after keepalives stabilise) with when devices reset after not receiving keepalives. Probably needs more work, especially around the first and/or second keepalives that vary significantly by device and firmware version. Review: https://reviewboard.asterisk.org/r/2611/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-09Use correct function for getting bridged peer when doing direct media checks.Mark Michelson
(closes issue ASTERISK-21947) reported by Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393897 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
2013-07-05Refactor 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
2013-07-03Revert accidental overcommit.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393632 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-03Add BUGBUG note for ASTERISK-22009Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-03chan_dahdi: Fix segfault reloading chan_dahdi when round robin is used.Richard Mudgett
* 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
2013-07-02Fix chan_gtalk.c compile error.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-02Fix issue with inability to cancell call transfer made by on-sceen menus.Igor Goncharovskiy
Reported by: Igor Olhovskiy ........ Merged revisions 393395 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393396 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
2013-06-28Add stasis publications for blind and attended transfers.Mark Michelson
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
2013-06-26Fix incorrect calls to ast_bridge_impart().Richard Mudgett
There was a misunderstanding about ast_bridge_impart()'s handling of the imparted channel's reference. The channel reference is passed by the caller unless ast_bridge_impart() returns an error. * Fixed a memory leak in conf_announce_channel_push() if the impart failed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-25Fix memory/ref counting leaks in a variety of locationsMatthew Jordan
This patch fixes the following memory leaks: * http.c: The structure containing the addresses to bind to was not being deallocated when no longer used * named_acl.c: The global configuration information was not disposed of * config_options.c: An invalid read was occurring for certain option types. * res_calendar.c: The loaded calendars on module unload were not being properly disposed of. * chan_motif.c: The format capabilities needed to be disposed of on module unload. In addition, this now specifies the default options for the maxpayloads and maxicecandidates in such a way that it doesn't cause the invalid read in config_options.c to occur. (issue ASTERISK-21906) Reported by: John Hardin patches: http.patch uploaded by jhardin (license 6512) named_acl.patch uploaded by jhardin (license 6512) config_options.patch uploaded by jhardin (license 6512) res_calendar.patch uploaded by jhardin (license 6512) chan_motif.patch uploaded by jhardin (license 6512) ........ Merged revisions 392810 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-22Merge in current pimp_my_sip work, including:Joshua Colp
1. Security events 2. Websocket support 3. Diversion header + redirecting support 4. An anonymous endpoint identifier 5. Inbound extension state subscription support 6. PIDF notify generation 7. One touch recording support (special thanks Sean Bright!) 8. Blind and attended transfer support 9. Automatic inbound registration expiration 10. SRTP support 11. Media offer control dialplan function 12. Connected line support 13. SendText() support 14. Qualify support 15. Inband DTMF detection 16. Call and pickup groups 17. Messaging support Thanks everyone! Side note: I'm reminded of the song "How Far We've Come" by Matchbox Twenty. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-22Migrate PeerStatus events to stasis, add stasis endpoints, and add ↵Joshua Colp
chan_pjsip device state. (closes issue ASTERISK-21489) (closes issue ASTERISK-21503) Review: https://reviewboard.asterisk.org/r/2601/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-21Add BUGBUG for broken direct media in chan_gulpMatthew Jordan
(issue ASTERISK-21947) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-21Change chan_unistim to use core transfer API.Mark Michelson
Review: https://reviewboard.asterisk.org/r/2553 (closes issue ASTERISK-21527) Reported by Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-17chan_vpb: Fix compile error and __ast_channel_alloc() prototype const ↵Richard Mudgett
inconsistency. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-17chan_misdn: Fix compile error after CDR merge.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-17Update Asterisk's CDRs for the new bridging frameworkMatthew Jordan
This patch is the initial push to update Asterisk's CDR engine for the new bridging framework. This patch guts the existing CDR engine and builds the new on top of messages coming across Stasis. As changes in channel state and bridge state are detected, CDRs are built and dispatched accordingly. This fundamentally changes CDRs in a few ways. (1) CDRs are now *very* reflective of the actual state of channels and bridges. This means CDRs track well with what an actual channel is doing - which is useful in transfer scenarios (which were previously difficult to pin down). It does, however, mean that CDRs cannot be 'fooled'. Previous behavior in Asterisk allowed for CDR applications, channels, and other properties to be spoofed in parts of the code - this no longer works. (2) CDRs have defined behavior in multi-party scenarios. This behavior will not be what everyone wants, but it is a defined behavior and as such, it is predictable. (3) The CDR manipulation functions and applications have been overhauled. Major changes have been made to ResetCDR and ForkCDR in particular. Many of the options for these two applications no longer made any sense with the new framework and the (slightly) more immutable nature of CDRs. There are a plethora of other changes. For a full description of CDR behavior, see the CDR specification on the Asterisk wiki. (closes issue ASTERISK-21196) Review: https://reviewboard.asterisk.org/r/2486/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-11Fix issue with no sound in both way in case of previous call to chan_unistim ↵Igor Goncharovskiy
phone was canceled. (related to ASTERISK-20183) ........ Merged revisions 391379 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-11IAX2: Transfer Reject: Lock bridgecallno before touching it, refactorAlec L Davis
1). When touching the bridgecallno, we need to lock it. 2). Remove magic number '0' and replace with TRANSFER_NONE. 3). Exit early if no bridgecallno. 4). Reduce indentation. Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2613/ ........ Merged revisions 391333 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 391334 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-10Change chan_skinny to use core transfer API.Damien Wedhorn
Changes for both attended and blind transfers in chan_skinny to use the new transfer API instead of masquerade. (closes issue ASTERISK-21526) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2557/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-10chan_iax2: nativebridge refactor, missed unlock bridgecallnoAlec L Davis
........ Merged revisions 391143 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 391148 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-10fix bad edit after conflict resolutionAlec L Davis
........ Merged revisions 391107 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 391111 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-10IAX2: refactor nativebridge transferAlec L Davis
remove triple checking of iaxs[fr->callno]->transferring reduce indentation. Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2602/ ........ Merged revisions 391065 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 391084 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-10IAX2: fix race condition with nativebridge transfers.Alec L Davis
1). When touching the bridgecallno, we need to lock it. 2). stop_stuff() which calls iax2_destroy_helper() Assumes the lock on the pvt is already held, when iax2_destroy_helper() is called. Thus we need to lock the bridgecallno pvt before we call stop_stuff(iaxs[fr->callno]->bridgecallno); 3). When evaluating the state of 'callno->transferring' of the current leg, we can't change it to READY unless the bridgecallno is locked. Why, if we are interrupted by the other call leg before 'transferring = TRANSFER_RELEASED', the interrupt will find that it is READY and that the bridgecallno is also READY so Releases the legs. (closes issue ASTERISK-21409) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2594/ ........ Merged revisions 391062 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 391063 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-07Refactor chan_dahdi/sig_analog/sig_pri and chan_misdn to use the common ↵Richard Mudgett
transfer functions. (closes issue ASTERISK-21523) Reported by: Matt Jordan (closes issue ASTERISK-21524) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2600/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-06Refactor the features configuration scheme.Mark Michelson
Features configuration is handled in its own API in features_config.h and features_config.c. This way, features configuration is accessible to anything that needs it. In addition, features configuration has been altered to be more channel-oriented. Most callers of features API code will be supplying a channel so that the individual channel's settings will be acquired rather than the global setting. Missing from this commit is XML documentation for the features configuration. That will be handled in a separate commit. Review: https://reviewboard.asterisk.org/r/2578/ (issue ASTERISK-21542) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-06Add a BUGBUG note.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-28Add attended transfer support for chan_sip.cMark Michelson
This now uses the core API for performing attended transfers. Review https://reviewboard.asterisk.org/r/2513 (Closes issue ASTERISK-21520) reported by Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-28Adds support for a core attended transfer function plus adds some hiding of ↵Mark Michelson
masquerades. The attended transfer API call can complete the attended transfer in a number of ways depending on the current bridged states of the channels involved. The hiding of masquerades is done in some bridging-related functions, such as the manager Bridge action and the Bridge dialplan application. In addition, call pickup was edited to "move" a channel rather than masquerade it. Review: https://reviewboard.asterisk.org/r/2511 (closes issue ASTERISK-21334) Reported by Matt Jordan (closes issue Asterisk-21336) Reported by Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-24Split Hold event into Hold/Unhold, and move it into core.Jason Parker
(closes issue ASTERISK-21487) Review: https://reviewboard.asterisk.org/r/2565/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-24Migrate a large number of AMI events over to Stasis-CoreMatthew Jordan
This patch moves a number of AMI events over to the Stasis-Core message bus. This includes: * ChanSpyStart/Stop * MonitorStart/Stop * MusicOnHoldStart/Stop * FullyBooted/Reload * All Voicemail/MWI related events In addition, it adds some Stasis-Core and AMI support for generic AMI messages, refactors the message router in AMI to use a single router with topic forwarding for the topics that AMI cares about, and refactors MWI message types and topics to be more name compliant. Review: https://reviewboard.asterisk.org/r/2532 (closes issue ASTERISK-21462) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-24Fix several problems caused by multiple line usage with i2004 phones.Igor Goncharovskiy
Reported by: Daniel Bohling, MihaiMircea (closes issue ASTERISK-21061) (closes issue ASTERISK-21120) ........ Merged revisions 389661 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-21Merge in the bridge_construction branch to make the system use the Bridging API.Richard Mudgett
Breaks many things until they can be reworked. A partial list: chan_agent chan_dahdi, chan_misdn, chan_iax2 native bridging app_queue COLP updates DTMF attended transfers Protocol attended transfers git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-19Add transfer softkey to ringout state to enable blond transfers.Damien Wedhorn
(closes issue ASTERISK-21327) Reported by: wedhorn Tested by: myself Patches: skinny-blindxfer01.diff uploaded by wedhorn (license 5019) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-18Add call forward no answer to skinny and cleanup general callfwd handling.Damien Wedhorn
CallforwardNoAnswer uses a sched to determine when to forward the call. Defaults to 20secs but configurable in skinny.conf. Adds dialType to each subchannel structure to be used to differentiate between normal dials that result in a call being placed (default) and other uses for the skinny_dialer (such as cfwd digit collection). Restructured all cfwd handling to use this new arrangement. (closes issue ASTERISK-21292) Reported by: wedhorn Tested by: myself Patches: skinny-callfwdnoans03.diff uploaded by wedhorn (license 5019) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-17Fix shutdown assertions in stasis-coreDavid M. Lee
In r388005, macros were introduced to consistently define message types. This added an assert if a message type was used either before it was initialized or after it had been cleaned up. It turns out that this assertion fires during shutdown. This actually exposed a hidden shutdown ordering problem. Since unsubscribing is asynchronous, it's possible that the message types used by the subscription could be freed before the final message of the subscription was processed. This patch adds stasis_subscription_join(), which blocks until the last message has been processed by the subscription. Since joining was most commonly done right after an unsubscribe, a stasis_unsubscribe_and_join() convenience function was also added. Similar functions were also added to the stasis_caching_topic and stasis_message_router, since they wrap subscriptions and have similar problems. Other code in trunk was refactored to join() where appropriate, or at least verify that the subscription was complete before being destroyed. Review: https://reviewboard.asterisk.org/r/2540 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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