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2013-07-25A great big renaming patchMatthew Jordan
This patch renames the bridging* files to bridge*. This may seem pedantic and silly, but it fits better in line with current Asterisk naming conventions: * channel is not "channeling" * monitor is not "monitoring" etc. A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is the act of using a bridge on a set of channels - and the API that fulfills that role is more than just the action. (closes issue ASTERISK-22130) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-25Improve initial INVITE handling and fix crash due to rapidly arriving CANCEL.Joshua Colp
(closes issue ASTERISK-22150) Review: https://reviewboard.asterisk.org/r/2696/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-23Drop the reference count on the correct object.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-23Reinclude sys/stat.h in chan_dahdi.c and remove redundant include in utils.cRichard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-23Some chan_dahdi protected function renaming.Richard Mudgett
analog_lib_handles --> dahdi_analog_lib_handles enable_dtmf_detect --> dahdi_dtmf_detect_enable disable_dtmf_detect --> dahdi_dtmf_detect_disable dahdi_enable_ec --> dahdi_ec_enable dahdi_disable_ec --> dahdi_ec_disable update_conf --> dahdi_conf_update dahdi_link --> dahdi_master_slave_link dahdi_unlink --> dahdi_master_slave_unlink (closes issue ASTERISK-22129) Reported by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-23Restore chan_dahdi native bridging and PRI tromboned call elimination.Richard Mudgett
Created a native_dahdi bridging technology for use with the new bridging API. The new bridging technology is part of the chan_dahdi channel driver because it is very specific to that driver. Rather than include the new code directly into chan_dahdi.c the new bridge technology is in its own file and linked into chan_dahdi.so. A large part of this change is the mechanical process of moving declarations around so chan_dahdi.c can be split up into more files later. * Changed the bridging core to pass NULL frames into the channel technologies instead of discarding them. The channel technologies may need the proding to determine if their configuration is still valid. (closes issue ASTERISK-21886) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2681/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-23Add DTLS-SRTP support to chan_pjsipKinsey Moore
This patch introduces DTLS-SRTP support to chan_pjsip and the options necessary to configure it including an option to allow choosing between 32 and 80 byte SRTP tag lengths. During the implementation and testing of this patch, three other bugs were found and their fixes are included with this patch. The two in chan_sip were a segfault relating to DTLS setup and mistaken call rejection. The third bug fix prevents chan_pjsip from attempting to perform bridge optimization between two endpoints if either of them is running any form of SRTP. Review: https://reviewboard.asterisk.org/r/2683/ (closes issue ASTERISK-21419) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-23Expose the chan_pjsip implementation pvt and session in a defined manner.Joshua Colp
This allows modules outside of chan_pjsip itself to get the session given only an Asterisk channel. Review: https://reviewboard.asterisk.org/r/2674/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-21Add additional control frame types to the IAX2 parser for debug messagesMatthew Jordan
This patch adds some of the more recent control frame types to the IAX2 parser. When IAX2 debugging is enabled, it will now show more of the control frame types. (closes issue ASTERISK-22120) Reported by: Birger "WIMPy" Harzenetter patches: iaxcmds.diff uploaded by wimpy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394895 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-20Add transfer support to CELKinsey Moore
This adds CEL support for blind and attended transfers and call pickup. During the course of adding this functionality I noticed that CONF_ENTER, CONF_EXIT, and BRIDGE_TO_CONF events are particularly useless without a bridge identifier, so I added that as well. This adds tests for blind transfers, several types of attended transfers, and call pickup. The extra field in CEL records now consists of a JSON blob whose fields are defined on a per-event basis. Review: https://reviewboard.asterisk.org/r/2658/ (closes issue ASTERISK-21565) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-19ARI: Bridge Playback, Bridge RecordJonathan Rose
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
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
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