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2012-01-13Remove some dead code in ast_bridge_call().Richard Mudgett
None of the parameters to ast_bridge_call() can be NULL for the bridge to work so no need to check for it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-12Adds peer to CEL report on CEL_BRIDGE_START and CEL_BRIDGE_ENDJonathan Rose
(closes issue ASTERISK-17940) Reporter: Nic Colledge Patches: features_18.patch uploaded by Nic Colledge (license 6245) ........ Merged revisions 350501 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350502 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09Replace direct access to channel name with accessor functionsTerry Wilson
There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-23INFO/Record request configurable to use dynamic featuresJonathan Rose
Adds two new options to SIP peers allowing them to specify features (dynamic or builtin) to use when sending INFO/record requests. Recordonfeature activates whatever feature is specified when recieving a record: on request while recordofffeature activates whatever feature is specified when receiving a record: off request. Both of these features can be disabled by setting the feature to an empty string. (closes issue ASTERISK-16507) Reported by: Jon Bright Review: https://reviewboard.asterisk.org/r/1634/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-16Clean-up on isle five for __ast_request_and_dial() and ast_call_forward().Richard Mudgett
* Add locking when a channel inherits variables and datastores in __ast_request_and_dial() and ast_call_forward(). Note: The involved channels are not active so there was minimal potential for problems. * Remove calls to ast_set_callerid() in __ast_request_and_dial() and ast_call_forward() because the set information is for the wrong direction. * Don't use C++ keywords for variable names in ast_call_forward(). * Run the redirecting interception macro if defined when forwarding a call in ast_call_forward(). Note: Currently will never execute because the only callers that supply a calling channel supply a hungup or zombie channel. * Make feature_request_and_dial() put the transferee into autoservice when it calls ast_call_forward() in case a redirection interception macro is run. Note: Currently will never happen because the caller channel (Party B) is always hungup at this time. * Make feature_request_and_dial() ignore the AST_CONTROL_PROCEEDING frame to silence a log message. ........ Merged revisions 348464 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348465 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-16Fix crash during CDR update.Richard Mudgett
The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to be called by different threads for the same channel. The channel driver thread and the PBX thread running dialplan. * Add lock protection around CDR API calls that access an ast_channel pointer. (closes issue ASTERISK-18836) Reported by: gpluser Review: https://reviewboard.asterisk.org/r/1628/ ........ Merged revisions 348362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348363 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-14Add and document PARKEDCALL variable set during timeoutJonathan Rose
PARKEDCALL variable tracks which parking lot the call was last parked in. This can be used afterwards for flow control when returntoorigin is set to off. I went ahead and documented both this and the existing variable set during timeout (PARKINGSLOT) in the sample features.conf since there was no prior mention of variables being set during timeout. (closes issue ASTERISK-16239) Reported By: Clod Patry Patches: M17503.diff uploaded by Clod Patry (license 5138) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-08Mark channel running the h exten with the soft-hangup flag.Richard Mudgett
When a bridge is broken, ast_bridge_call() might execute the h exten on the calling channel. However, that channel may not have been the channel that broke the bridge by hanging up. The channel executing the h exten must be in a hung up state so things like AGI run in the correct mode. * Make sure ast_bridge_call() marks the channel it is executing the h exten on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as to match the pbx.c main dialplan execution loop when it executes the h exten.) (closes issue ASTERISK-18811) Reported by: David Hajek Patches: jira_asterisk_18811_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: David Hajek, rmudgett ........ Merged revisions 347595 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347600 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347601 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-24Fixes a segfault caused by referencing null frames introduced in r338623Jonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-20Fix AGI exec Park to honor the Park application parameters.Richard Mudgett
The fix for ASTERISK-12715 and ASTERISK-12685 added a check for the Park application because the channel needed to be masqueraded to prevent a crash. Since the Park application now always masquerades the channel into the parking lot, the special check is no longer needed. The fix also resulted in AGI exec Park attempting to double park the call and not honor the Park application parameters. * Removed no longer necessary call to ast_masq_park_call() by AGI exec for the Park application. (Reverts -r146923) * Fix Park application to only return 0 or -1. The AGI exec Park was causing broken pipe error messages because the Park application returned 1 on successful park. (closes issue ASTERISK-18737) ........ Merged revisions 341717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341718 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341719 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-18More parking issues.Richard Mudgett
* Fix potential deadlocks in SIP and IAX blind transfer to parking. * Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the parkext_exclusive option with transfers (Park(,,,,,exclusive_lot) parameter). Created ast_park_call_exten() and ast_masq_park_call_exten() to maintian API compatibility. * Made masq_park_call() handle a failed ast_channel_masquerade() setup. * Reduced excessive struct parkeduser.peername[] size. ........ Merged revisions 341254 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341255 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-13Fix DTMF blind transfer continuing to execute dialplan after transfer.Richard Mudgett
Party A calls Party B. Party A DTMF blind transfers Party B to Party C. Party A channel continues to execute dialplan. * Fixed the return value of builtin_blindtransfer() to return the correct value after a transfer so the dialplan will not keep executing. * Removed unnecessary connected line update that did not really do anything. * Made access to GOTO_ON_BLINDXFR thread safe in check_goto_on_transfer(). * Fixed leak of xferchan for failure cases in check_goto_on_transfer(). * Updated debug messages in builtin_blindtransfer() and check_goto_on_transfer(). (closes issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 340809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340810 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340813 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-30Preserve DTMF length in main/features.cOlle Johansson
Review: https://reviewboard.asterisk.org/r/1463/ A small part of much larger work with DTMF duration in Asterisk, funded by IPvision AS in Denmark. Thanks to irroot for the review! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-09Merged revisions 335078 via svnmerge from Matthew Jordan
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines Merged revisions 335064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-07Merged revisions 334841 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334841 | rmudgett | 2011-09-07 14:33:38 -0500 (Wed, 07 Sep 2011) | 17 lines Merged revisions 334840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334840 | rmudgett | 2011-09-07 14:31:44 -0500 (Wed, 07 Sep 2011) | 10 lines Fix AMI action Park crash. * Made AMI action Park not say anything to the parker channel (AMI header Channel2) since the AMI action is a third party parking the call. (This is a change in behavior that cannot be preserved without a lot of effort.) * Made not play pbx-parkingfailed if the Park 's' option is used. JIRA AST-660 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-07Merged revisions 334747 via svnmerge from Stefan Schmidt
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334747 | schmidts | 2011-09-07 15:10:37 +0000 (Wed, 07 Sep 2011) | 9 lines Merged revisions 334682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011) | 3 lines Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-07clean up wrong merged stuff Stefan Schmidt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-07Merged revisions 334682 via svnmerge from Stefan Schmidt
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011) | 3 lines Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334712 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-07Adding the Feature to sent a Reason Header in a SIP Cancel message by set ↵Stefan Schmidt
the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-16Merged revisions 332101 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332101 | rmudgett | 2011-08-16 12:17:28 -0500 (Tue, 16 Aug 2011) | 140 lines Merged revisions 332100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332100 | rmudgett | 2011-08-16 11:31:36 -0500 (Tue, 16 Aug 2011) | 133 lines Fix multiple parking issues. JIRA ASTERISK-17183 Multi-parkinglot directs calls to wrong parkinglot. JIRA ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430 ParkedCall() with no extension should pickup first available call and does not. JIRA AST-576 Issues with parking lots * Removed searching for parking lots by extension. Parking lots can only be found by the parking lot name since parking lot access extensions and spaces are not guaranteed to be unique. * Added parking_lot_name option to the Park and ParkedCall applications. Updated documentation for Park and ParkedCall applications. * Add parkext_exclusive configuration option to make parking entry extensions specify which parking lot they access. (closes issue ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett, David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi Quezada (closes issue ASTERISK-17430) Reported by: Philippe Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA AST-624 'next' setting for findslot does nothing * Reimplemented since findslot feature option broken by -r114655. (closes issue ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett JIRA ASTERISK-15792 Dialplan continues execution after transfer to park. This happens for DTMF attended transfer, DTMF blind transfer, and DTMF one-touch-parking if the party initiating these features also initiated the call. * Fixed the return code from the affected builtin features when parking a call. (closes issue ASTERISK-15792) Reported by: Mat Murdock Tested by: rmudgett, twilson JIRA AST-607 The courtesytone is not playing to the expected call when picking up a parked call. This is mostly a documentation problem. However, the option is not reset to the default when features.conf is reloaded. * Updated features.conf.sample documentation for courtesytone and parkedplay options. * Reset the parkedplay option to default when features.conf is reloaded. JIRA AST-615 AMI Park action followed by features reload results in orphaned channels in parking lot. * Reloading features.conf will not touch parking lots that have calls still parked in them. Reload again at a later time. Misc additional fixes: * Added unit test for parking lot dialplan usage checking. * Made update connected line when a parked call is retrieved from a parking lot. * Made retrieved parked call stop ringing or MOH depending upon how the call was waiting in the parking lot. * Made CLI "features show" indicate if the parking lot is enabled for use. * Added PARKINGDYNEXTEN channel variable to allow dynamic parking lots to specify the parking lot access extension. * Made AMI ParkedCalls action ParkedCall events have a Parkinglot header. * Made AMI ParkedCalls action ParkedCallsComplete event have a Total header. * Fixed potential deadlock from AMI Park action holding channel locks while calling masq_park_call(). * Fixed several places where ast_strdupa() were used inside of loops. (Mostly fixed by refactoring the loop body into its own function.) * Fixed copy_parkinglot() copying too much from the source parking lot. Extracted the parking lot configuration settings into struct parkinglot_cfg. * Refactored courtesytone playing code to put the channel not playing the tone in autoservice. * Fix when pbx-parkingfailed is played that the other channel is put in autoservice if it exists. * Fixed parkinglot reference leak in parked_call_exec() error paths. * Fixed parkinglot_unref() use of parkinglot after it was unreffed. * Made destroy the struct ast_parkinglot parkings lock when done. * Refactored the features.conf parking lot configuration code to eliminate redundancy. * Fixed feature reload to better protect parking lots. * Fixed parking lot container reference leak in handle_parkedcalls(). * Fixed the total count in handle_parkedcalls(). Review: https://reviewboard.asterisk.org/r/1358/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-10Merged revisions 331420 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331420 | rmudgett | 2011-08-10 14:07:53 -0500 (Wed, 10 Aug 2011) | 2 lines Make sure feature_request_and_dial() initializes outstate if passed in. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-10Merged revisions 331418 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331418 | rmudgett | 2011-08-10 13:25:08 -0500 (Wed, 10 Aug 2011) | 6 lines Revert -r318141. It was a band-aid that only partially fixed parking. A better fix is on reviewboard review 1358. (issue ASTERISK-17374) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-09Merged revisions 331265 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331265 | rmudgett | 2011-08-09 18:12:49 -0500 (Tue, 09 Aug 2011) | 22 lines Merged revisions 331248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331248 | rmudgett | 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011) | 15 lines Misc minor items found in code. * Add some reentrancy protection in pbx.c when creating the contexts_table hash table. * Fix inverted test in chan_sip.c conditional code. * Fix uninitialized variable and use of the wrong variable in chan_iax2.c. * Fix test of return value in app_parkandannounce.c. Explicitly testing for -1 is bad if the function does not actually return that value when it fails. * Fixup some comments and add some curly braces in features.c. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-21Merged revisions 329257 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r329257 | russell | 2011-07-21 15:22:36 -0500 (Thu, 21 Jul 2011) | 2 lines s/1.10/10.0/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-21Merged revisions 329145 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329145 | rmudgett | 2011-07-21 11:52:17 -0500 (Thu, 21 Jul 2011) | 16 lines Merged revisions 329144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329144 | rmudgett | 2011-07-21 11:46:21 -0500 (Thu, 21 Jul 2011) | 9 lines Dialplan bridge() app mutex 'current_dest_chan' freed more times than we've locked! This appears to be a leftover from when ast_channel was converted to ao2 objects. Simply removed the extraneous unlock. (closes issue ASTERISK-17772) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-14Merged revisions 328247 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-23Merged revisions 324652 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r324652 | dvossel | 2011-06-23 13:23:21 -0500 (Thu, 23 Jun 2011) | 20 lines Merged revisions 324634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r324634 | dvossel | 2011-06-23 13:18:46 -0500 (Thu, 23 Jun 2011) | 13 lines Merged revisions 324627 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011) | 7 lines Addresses AST-2011-010, remote crash in IAX2 driver Thanks to twilson for identifying the issue and providing the patches. AST-2011-010 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-15Merged revisions 323754 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323754 | twilson | 2011-06-15 13:21:52 -0500 (Wed, 15 Jun 2011) | 23 lines Merged revisions 323733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323733 | twilson | 2011-06-15 13:13:00 -0500 (Wed, 15 Jun 2011) | 16 lines Merged revisions 323732 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011) | 9 lines Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were broken by a recent DTMF change. This patch makes sure that dynamic features are also checked when deciding whether or not to pass DTMF through or store it for interpreting. (closes issue ASTERISK-17914) Reported by: vrban ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-09Merged revisions 322749 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322749 | rmudgett | 2011-06-09 11:31:53 -0500 (Thu, 09 Jun 2011) | 15 lines Remove potential deadlock in call pickup race. Deadlock is possible in ast_do_pickup() when holding the target channel lock and trying to get the chan channel lock. Also, holding the target lock when calling ast_channel_masquerade() is not a good idea because that routine does deadlock avoidance. * Removed the need to hold the target lock after marking the target with a datastore and getting the connected line data off of the target channel. * Moved can_pickup() to ast_can_pickup() in features.c. Now all the call pickup methods use the same basic call pickup availability check. Review: https://reviewboard.asterisk.org/r/1234/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-27Merged revisions 321333 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321333 | lmadsen | 2011-05-27 17:40:23 -0400 (Fri, 27 May 2011) | 7 lines Allow parking lot hints and musicclass to be set. (closes issue #19378) Reported by: sboily_proformatique Patches: pf_parkinghint_music_fix uploaded by sboily proformatique (license 206) Tested by: russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-27Merged revisions 321211 via svnmerge from Alec L Davis
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321211 | alecdavis | 2011-05-27 20:31:15 +1200 (Fri, 27 May 2011) | 16 lines Fix *8 directed pickup locks system during pickupsound play out move playout from sip_pickup_thread to bridge using BRIDGE_PLAY_SOUND method, This stop the clash of 2 threads trying to write audio to same channel. In addition fixes choppy audio beep in issue 19177. (issue #18654) (issue #19177) Reported by: Docent Patches: review1232-1.8.diff.txt alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1232/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-25Merged revisions 320823 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines The AMI Newstate event contains different information between v1.4 and v1.8. The addition of connected line support in v1.8 changes the behavior of the channel caller ID somewhat. The channel caller ID value no longer time shares with the connected line ID on outgoing call legs. The timing of some AMI events/responses output the connected line ID as caller ID. These party ID's are now separate. * The ConnectedLineNum and ConnectedLineName headers were added to many AMI events/responses if the CallerIDNum/CallerIDName headers were also present. (closes issue #18252) Reported by: gje Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1227/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-25Merged revisions 320796 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320796 | rmudgett | 2011-05-25 11:23:11 -0500 (Wed, 25 May 2011) | 17 lines Give zombies a safe channel driver to use. Recent crashes from zombie channels suggests that they need a safe home to goto. When a masquerade happens, the physical part of the zombie channel is hungup. The hangup normally sets the channel private pointer to NULL. If someone then blindly does a callback to the channel driver, a crash is likely because the private pointer is NULL. The masquerade now sets the channel technology of zombie channels to the kill channel driver. Related to the following issues: (issue #19116) (issue #19310) Review: https://reviewboard.asterisk.org/r/1224/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-20Merged revisions 320059 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320059 | rmudgett | 2011-05-20 12:03:49 -0500 (Fri, 20 May 2011) | 1 line Misc comment cleanup in features.c. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-20Merged revisions 320057 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320057 | rmudgett | 2011-05-20 11:43:02 -0500 (Fri, 20 May 2011) | 19 lines Crash while transferring a call during DTMF feature timeout. When a call is being attended transferred during the time between AST_FRAME_DTMF_BEGIN and AST_FRAME_DTMF_END, the transferred channel becomes a zombie (so tech data is not available), making ast_dtmf_stream() segfault when it tries to send the DTMF digit (at least with SIP channels). Patch based on feature-end-zombie.patch uploaded by Irontec (license 1256) * Check for zombies when ast_channel_bridge() returns. * Guarantee that the fo parameter value is initialized in ast_channel_bridge() before any returns. (closes issue #19116) Reported by: Irontec Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320058 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-20Merged revisions 320007 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320007 | rmudgett | 2011-05-20 11:19:01 -0500 (Fri, 20 May 2011) | 2 lines Change some variable names to make pickup code easier to understand. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-20Merged revisions 319997 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319997 | rmudgett | 2011-05-20 10:48:25 -0500 (Fri, 20 May 2011) | 25 lines Crash when using directed pickup applications. The directed pickup applications can cause a crash if the pickup was successful because the dialplan keeps executing. This patch does the following: * Completes the channel masquerade on a successful pickup before the application returns. The channel is now guaranteed a zombie and must not continue executing the dialplan. * Changes the return value of the directed pickup applications to return zero if the pickup failed and nonzero(-1) if the pickup succeeded. * Made some code optimizations that no longer require re-checking the pickup channel to see if it is still available to pickup. (closes issue #19310) Reported by: remiq Patches: issue19310_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, remiq, rmudgett Review: https://reviewboard.asterisk.org/r/1221/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-19Merged revisions 319866 via svnmerge from Jonathan Rose
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319866 | jrose | 2011-05-19 13:32:38 -0500 (Thu, 19 May 2011) | 11 lines Fix Randomize option on Park() The randomize option was generally not working like it should have at all on Park(). This patch restores intended functionality. (closes issue #18862) Reported by: davidw Tested by: jrose Review: https://reviewboard.asterisk.org/r/1222/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319867 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-13Merged revisions 318868 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318868 | rmudgett | 2011-05-13 11:28:26 -0500 (Fri, 13 May 2011) | 19 lines CDR's are being written immediately on caller hangup. CDR's are being written immediately on caller hangup. The dialplan is not able to modify it in the h exten. The h exten in the initial context is not run before closing CDR's when the bridge is unlinked if a macro is active and does not have an h exten. * Make ast_bridge_call() check for an h exten in the current context and if a macro is active then the initial context. The first h exten found is then run before closing the CDR. (closes issue #18212) Reported by: leearcher Patches: issue18212_v1.8.patch uploaded by rmudgett (license 664) Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1206/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-12Merged revisions 318671 via svnmerge from Alec L Davis
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines Fix directed group pickup feature code *8 with pickupsounds enabled Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues. 1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked. 2). dialplan applications for directed_pickups shouldn't beep. 3). feature code for directed pickup should beep on success/failure if configured. Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite. Moved app_directed:pickup_do() to features:ast_do_pickup(). Functions below, all now use the new ast_do_pickup() app_directed_pickup.c: pickup_by_channel() pickup_by_exten() pickup_by_mark() pickup_by_part() features.c: ast_pickup_call() (closes issue #18654) Reported by: Docent Patches: ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett Review: https://reviewboard.asterisk.org/r/1185/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-09Merged revisions 318282 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318282 | rmudgett | 2011-05-09 14:07:01 -0500 (Mon, 09 May 2011) | 24 lines Hangup extension executed twice. When a user hangs up a call, in certain circumstances, the hangup extension can end up being executed twice: 1) If a call is bridged and the 'h' extension executes the Hangup application, then the 'h' extension will be executed twice. 2) If a call is bridged within a macro (Dial or Queue), it has its own 'h' extension, the main context also has an 'h' extension, and the macro 'h' extension executes the Hangup application, then both 'h' extensions will be executed. * Revert originally commited fix for #16106 and just set AST_FLAG_BRIDGE_HANGUP_RUN unconditionally in ast_bridge_call(). The bridge code just executed an 'h' extension so the main PBX loop does not need to execute one as well. (issue #16106) Reported by: ajohnson (issue #16548) Reported by: hajekd ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-09Minor change to 318141 to improve parsing behavior.Jonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318193 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-09Allows ParkedCall application to specify a parkinglot.Jonathan Rose
When invoking the app parkedcall, the argument can now include '@parkinglot' after the extension. (closes issue #18777) Reported by: cartama Patches: 0018777.diff uploaded by cartama (license 1157) Review: https://reviewboard.asterisk.org/r/1209/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-03Merged revisions 316265 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines Fix a bunch of compiler warnings generated by gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-26Merged revisions 315644 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315644 | twilson | 2011-04-26 14:39:01 -0700 (Tue, 26 Apr 2011) | 32 lines Merged revisions 315643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315643 | twilson | 2011-04-26 14:27:44 -0700 (Tue, 26 Apr 2011) | 25 lines Merged revisions 315596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) | 18 lines Allow transfer loops without allowing forwarding loops We try to avoid the situation where two phones may be forwarded to each other causing an infinite loop by storing each dialed interface in a channel datastore and checking the list before dialing out. This works, but currently breaks situations like A calls B, A transfers B to C, B transfers C to A, and A transfers C to B. Since human interaction is happening here and not an automated forwarding loop, it should be allowed. This patch removes the dialed_interfaces datastore when a call is bridged (a suggestion from the brilliant mmichelson). If a call is being bridged, it should be safe to assume that we aren't stuck in a loop. Since we are now handling this is the bridge code, the previous attempts at handling it in app_dial and app_queue are removed. Review: https://reviewboard.asterisk.org/r/1195/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-07Merged revisions 313048 via svnmerge from Jonathan Rose
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313048 | jrose | 2011-04-07 08:35:33 -0500 (Thu, 07 Apr 2011) | 16 lines Merged revisions 313047 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r313047 | jrose | 2011-04-07 08:23:01 -0500 (Thu, 07 Apr 2011) | 9 lines Makes parking lots clear and rebuild properly when features reload is invoked from CLI Before, default parkinglot in context parkedcalls with ext 700 would always be present and when reload was invoked, the previous parkinglots would not be cleared. (closes issue #18801) Reported by: mickecarlsson Review: https://reviewboard.asterisk.org/r/1161/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313049 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-16Merged revisions 310902 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310902 | twilson | 2011-03-16 12:19:57 -0500 (Wed, 16 Mar 2011) | 43 lines Merged revisions 310889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310889 | twilson | 2011-03-16 12:03:27 -0500 (Wed, 16 Mar 2011) | 36 lines Merged revisions 310888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011) | 29 lines Don't delay DTMF in core bridge while listening for DTMF features This patch is mostly the work of Olle Johansson. I did some cleanup and added the silence generating code if transmit_silence is set. When a channel listens for DTMF in the core bridge, the outbound DTMF is not sent until we have received DTMF_END. For a long DTMF, this is a disaster. We send 4 seconds of DTMF to Asterisk, which sends no audio for those 4 seconds. Some products see this delay and the time skew on RTP packets that results and start ignoring the audio that is sent afterward. With this change, the DTMF_BEGIN frame is inspected and checked. If it matches a feature code, we wait for DTMF_END and activate the feature as before. If transmit_silence=yes in asterisk.conf, silence is sent if we paritally match a multi-digit feature. If it doesn't match a feature, the frame is forwarded along with the DTMF_END without delay. By doing it this way, DTMF is not delayed. (closes issue #15642) Reported by: jasonshugart Patches: issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license 396) Tested by: globalnetinc, jde (closes issue #16625) Reported by: sharvanek Review: https://reviewboard.asterisk.org/r/1092/ Review: https://reviewboard.asterisk.org/r/1125/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310941 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-09Allow parkedmusicclass to be settable for non-default parking lots.Jeff Peeler
(closes issue #17946) Reported by: bluecrow76 Patches: asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-09Merged revisions 307228 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307228 | jpeeler | 2011-02-09 13:52:51 -0600 (Wed, 09 Feb 2011) | 17 lines Merged revisions 307227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r307227 | jpeeler | 2011-02-09 13:52:12 -0600 (Wed, 09 Feb 2011) | 11 lines Make sure to set parking dial context for non-default parking lots. Since parking_con_dial isn't settable, set all parking lots to "park-dial". (closes issue #17946) Reported by: bluecrow76 Patches: asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by bluecrow76 (license 270) modified by me ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-07Pass a MCID request to the bridged channel.Richard Mudgett
Pass a MCID request to the bridged channel so the bridged channel can send it to the network. The ability to send the MCID request on an ISDN span is enabled with the new chan_dahdi.conf mcid_send option. JIRA SWP-2845 JIRA ABE-2736 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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