asterisk

Commit Graph

Author	SHA1	Message	Date
Jonathan Rose	19a9761084	Adds peer to CEL report on CEL_BRIDGE_START and CEL_BRIDGE_END (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	14 years ago
Terry Wilson	04da92c379	Replace direct access to channel name with accessor functions 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	14 years ago
Jonathan Rose	19a4928fee	INFO/Record request configurable to use dynamic features 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	14 years ago
Richard Mudgett	be74e6f16e	Clean-up on isle five for __ast_request_and_dial() and ast_call_forward(). * 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	14 years ago
Richard Mudgett	b05d4603c4	Fix crash during CDR update. 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	14 years ago
Jonathan Rose	480d46f92c	Add and document PARKEDCALL variable set during timeout 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	14 years ago
Richard Mudgett	3f13a41886	Mark channel running the h exten with the soft-hangup flag. 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	14 years ago
Jonathan Rose	f6cd5af36b	Fixes a segfault caused by referencing null frames introduced in r338623 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b961d57c4c	Fix AGI exec Park to honor the Park application parameters. 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	14 years ago
Richard Mudgett	10de040b6e	More parking issues. * 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	14 years ago
Richard Mudgett	cabf08b511	Fix DTMF blind transfer continuing to execute dialplan after transfer. 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	14 years ago
Olle Johansson	6e0f961432	Preserve DTMF length in main/features.c 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	14 years ago
Matthew Jordan	8b5ba33fe0	Merged revisions 335078 via svnmerge from 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	14 years ago
Richard Mudgett	3d63ec89e0	Merged revisions 334841 via svnmerge from 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	14 years ago
Stefan Schmidt	081dcb4a46	Merged revisions 334747 via svnmerge from 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	14 years ago
Stefan Schmidt	40f505c009	clean up wrong merged stuff git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	334401e57d	Merged revisions 334682 via svnmerge from 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	14 years ago
Stefan Schmidt	e549520b78	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@334683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	3ad6dccac8	Merged revisions 332101 via svnmerge from 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	14 years ago
Richard Mudgett	fa794d8f7a	Merged revisions 331420 via svnmerge from 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	14 years ago
Richard Mudgett	02ecb12f64	Merged revisions 331418 via svnmerge from 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	14 years ago
Richard Mudgett	b99b1116be	Merged revisions 331265 via svnmerge from 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	14 years ago
Russell Bryant	f243d129c9	Merged revisions 329257 via svnmerge from 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	14 years ago
Richard Mudgett	3b80737787	Merged revisions 329145 via svnmerge from 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	14 years ago
Leif Madsen	a525edea59	Merged revisions 328247 via svnmerge from 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	14 years ago
David Vossel	d5ea9e5ae2	Merged revisions 324652 via svnmerge from 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	14 years ago
Terry Wilson	c33e1b0e27	Merged revisions 323754 via svnmerge from 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	14 years ago
Richard Mudgett	0a8f9d2cf0	Merged revisions 322749 via svnmerge from 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	14 years ago
Leif Madsen	a2ca0997a6	Merged revisions 321333 via svnmerge from 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	14 years ago
Alec L Davis	7cc83a9018	Merged revisions 321211 via svnmerge from 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	14 years ago
Richard Mudgett	0096238b52	Merged revisions 320823 via svnmerge from 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	14 years ago
Richard Mudgett	a42bf8cc92	Merged revisions 320796 via svnmerge from 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	14 years ago
Richard Mudgett	2af231dd91	Merged revisions 320059 via svnmerge from 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	14 years ago
Richard Mudgett	ae091d166a	Merged revisions 320057 via svnmerge from 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	14 years ago
Richard Mudgett	b1cfd0e059	Merged revisions 320007 via svnmerge from 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	14 years ago
Richard Mudgett	0436c501c9	Merged revisions 319997 via svnmerge from 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	14 years ago
Jonathan Rose	87004f0d9f	Merged revisions 319866 via svnmerge from 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	14 years ago
Richard Mudgett	db89abf0bd	Merged revisions 318868 via svnmerge from 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	14 years ago
Alec L Davis	892b7a2efd	Merged revisions 318671 via svnmerge from 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	14 years ago
Richard Mudgett	bf57bb3c89	Merged revisions 318282 via svnmerge from 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	14 years ago
Jonathan Rose	ff4c7d46c0	Minor change to 318141 to improve parsing behavior. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318193 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	229e066dcb	Allows ParkedCall application to specify a parkinglot. 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	14 years ago
Russell Bryant	37aa52fd78	Merged revisions 316265 via svnmerge from 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	14 years ago
Terry Wilson	8d2a71877a	Merged revisions 315644 via svnmerge from 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	14 years ago
Jonathan Rose	68dd87ef0d	Merged revisions 313048 via svnmerge from 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	14 years ago
Terry Wilson	d958ca6953	Merged revisions 310902 via svnmerge from 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	14 years ago
Jeff Peeler	e2df246636	Allow parkedmusicclass to be settable for non-default parking lots. (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	15 years ago
Jeff Peeler	6b0fa46103	Merged revisions 307228 via svnmerge from 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	15 years ago
Richard Mudgett	49feb747ba	Pass a MCID request to the bridged channel. 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	15 years ago
Terry Wilson	1277a80a5b	Merged revisions 306674 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306674 \| twilson \| 2011-02-07 14:43:22 -0800 (Mon, 07 Feb 2011) \| 24 lines Merged revisions 306673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306673 \| twilson \| 2011-02-07 14:40:20 -0800 (Mon, 07 Feb 2011) \| 17 lines Merged revisions 306672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306672 \| twilson \| 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011) \| 10 lines Don't try to pickup a call in the middle of a masquerade If A calls B which doesn't answer and C & D both try to do a call pickup, it is possible for ast_pickup_call to answer the call, then fail to masquerade one of the calls because the other one is already in the process of masquerading. This patch checks to see if the channel is in the process of masquerading before call before selecting it for a pickup. Review: https://reviewboard.asterisk.org/r/1094/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	3556e4c2d4	Replace ast_log(LOG_DEBUG, ...) with ast_debug() (closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	fed10ed35d	Merged revisions 306124 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306124 \| jpeeler \| 2011-02-03 14:50:48 -0600 (Thu, 03 Feb 2011) \| 17 lines Merged revisions 306123 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r306123 \| jpeeler \| 2011-02-03 14:49:48 -0600 (Thu, 03 Feb 2011) \| 10 lines Set exception on channel in parking thread when POLLPRI event detected. This is done just to make the code be equivalent to the old select code. As noted in 303106 the same issue was already fixed in this branch, but the exception was not set on the channel in the case of POLLPRI. The reason that this did not cause a problem here is because in 122923 the check in __ast_read to check the exception flag was removed. (related to #18637) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306125 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	c26c190711	Asterisk media architecture conversion - no more format bitfields This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	8677f0424e	Merged revisions 304339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304339 \| jpeeler \| 2011-01-26 16:27:30 -0600 (Wed, 26 Jan 2011) \| 9 lines Merged revisions 304338 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304338 \| jpeeler \| 2011-01-26 16:26:37 -0600 (Wed, 26 Jan 2011) \| 2 lines Change delimiter used internally for GOTO_ON_BLINDXFR to commas to match 76703. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304340 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ca014f49a2	Merged revisions 304007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304007 \| rmudgett \| 2011-01-25 17:28:25 -0600 (Tue, 25 Jan 2011) \| 22 lines Merged revisions 304006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304006 \| rmudgett \| 2011-01-25 17:25:32 -0600 (Tue, 25 Jan 2011) \| 15 lines Merged revisions 304005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304005 \| rmudgett \| 2011-01-25 17:21:09 -0600 (Tue, 25 Jan 2011) \| 8 lines DTMF attended transfers sometimes fail for no apparent reason. The loop in feature_request_and_dial() can exit when Party C has answered without processing an AST_CONTROL_ANSWER. Also sometimes an AST_CONTROL_ANSWER never happens even though Party C has answered. Don't hangup Party C if he is up or we receive an AST_CONTROL_ANSWER. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	092134399c	Merged revisions 303549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303549 \| russell \| 2011-01-24 14:51:37 -0600 (Mon, 24 Jan 2011) \| 45 lines Merged revisions 303548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303548 \| russell \| 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) \| 38 lines Merged revisions 303546 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303546 \| russell \| 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) \| 31 lines Fix channel redirect out of MeetMe() and other issues with channel softhangup. Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped working properly. This issue includes a patch that resolves the issue by removing a call to ast_check_hangup() from app_meetme.c. I left that in my patch, as it doesn't need to be there. However, the rest of the patch fixes this problem with or without the change to app_meetme. The key difference between what happens before and after this patch is the effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(), ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme sees this which causes it to exit as intended. Checking ast_check_hangup() caused app_meetme to exit earlier in the process, and the target of the redirect saw the condition where ast_read() returned NULL. Removing ast_check_hangup() works around the issue in app_meetme, but doesn't solve the issue if another application did the same thing. There are also other edge cases where if an application finishes at the same time that a redirect happens, the target of the redirect will think that the channel hung up. So, I made some changes in pbx.c to resolve it at a deeper level. There are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to abort the hangup process. My patch extends this to remove the END_OF_Q frame from the channel's read queue, making the "abort hangup" more complete. This same technique was used in every place where a softhangup flag was cleared. (closes issue #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: https://reviewboard.asterisk.org/r/1082/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Shaun Ruffell	80f6848ca3	Merged revisions 303107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303107 \| sruffell \| 2011-01-20 13:57:31 -0600 (Thu, 20 Jan 2011) \| 23 lines Merged revisions 303106 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r303106 \| sruffell \| 2011-01-20 13:56:34 -0600 (Thu, 20 Jan 2011) \| 15 lines main/features: Use POLLPRI when waiting for events on parked channels. This change resolves a regression in the 1.6.2 when converting from select to poll. The DAHDI timers use POLLPRI to indicate that the timer fired, but features was not waiting for that flag. The result was no audio for MOH when a call was parked and res_timing_dahdi was in use. This patch is slightly modified from the one on the mantis issue. It does not set an exception on the channel if the POLLPRI flag is set. (closes issue #18262) Reported by: francesco_r Patches: patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029) Tested by: francesco_r, rfrantik, one47 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	c8e57f82bf	Merged revisions 302713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302713 \| rmudgett \| 2011-01-19 15:29:22 -0600 (Wed, 19 Jan 2011) \| 29 lines Merged revisions 302693 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302693 \| rmudgett \| 2011-01-19 15:25:41 -0600 (Wed, 19 Jan 2011) \| 22 lines Merged revisions 302671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302671 \| rmudgett \| 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011) \| 15 lines DTMF transfer plays the wrong sounds for wrong number or other call failure. * Set the default for features.conf.sample xferfailsound option to "beeperr" as documented instead of "pbx-invalid" and corrected the use of it in DTMF blind transfer (#1). * Improved DTMF blind transfer handling of wrong numbers. Most of the concerns in this issue were taken care of by the patch for issue 17999: Issues with DTMF triggered attended transfers. (closes issue #18379) Reported by: gincantalupo Tested by: rmudgett ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	f4d63bf918	Merged revisions 302552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302552 \| seanbright \| 2011-01-19 13:54:47 -0500 (Wed, 19 Jan 2011) \| 14 lines Merged revisions 302551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302551 \| seanbright \| 2011-01-19 13:54:03 -0500 (Wed, 19 Jan 2011) \| 7 lines Remove an extraneous \r\n at the end of a parking manager events. (closes issue #18363) Reported by: clegall_proformatique Patches: asterisk_1.8_295998_parking_manager_events_format.patch uploaded by clegall proformatique (license 1139) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	8cd1ac534b	Merged revisions 302318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302318 \| rmudgett \| 2011-01-18 16:04:14 -0600 (Tue, 18 Jan 2011) \| 1 line Use the expanded format type instead of plain int. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a05aeff312	Merged revisions 302174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302174 \| rmudgett \| 2011-01-18 12:11:43 -0600 (Tue, 18 Jan 2011) \| 102 lines Merged revisions 302173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302173 \| rmudgett \| 2011-01-18 12:07:15 -0600 (Tue, 18 Jan 2011) \| 95 lines Merged revisions 302172 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302172 \| rmudgett \| 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011) \| 88 lines Issues with DTMF triggered attended transfers. Issue #17999 1) A calls B. B answers. 2) B using DTMF dial 2 (code in features.conf for attended transfer). 3) A hears MOH. B dial number C 4) C ringing. A hears MOH. 5) B hangup. A still hears MOH. C ringing. 6) A hangup. C still ringing until "atxfernoanswertimeout" expires. For v1.4 C will ring forever until C answers the dead line. (Issue #17096) Problem: When A and B hangup, C is still ringing. Issue #18395 SIP call limit of B is 1 1. A call B, B answered 2. B 2(atxfer) call C 3. B hangup, C ringing 4. Timeout waiting for C to answer 5. Recall to B fails because B has reached its call limit. Because B reached its call limit, it cannot do anything until the transfer it started completes. Issue #17273 Same scenario as issue 18395 but party B is an FXS port. Party B cannot do anything until the transfer it started completes. If B goes back off hook before C answers, B hears ringback instead of the expected dialtone. ******** Note for the issue #17273 and #18395 fix: DTMF attended transfer works within the channel bridge. Unfortunately, when either party A or B in the channel bridge hangs up, that channel is not completely hung up until the transfer completes. This is a real problem depending upon the channel technology involved. For chan_dahdi, the channel is crippled until the hangup is complete. Either the channel is not useable (analog) or the protocol disconnect messages are held up (PRI/BRI/SS7) and the media is not released. For chan_sip, a call limit of one is going to block that endpoint from any further calls until the hangup is complete. For party A this is a minor problem. The party A channel will only be in this condition while party B is dialing and when party B and C are conferring. The conversation between party B and C is expected to be a short one. Party B is either asking a question of party C or announcing party A. Also party A does not have much incentive to hangup at this point. For party B this can be a major problem during a blonde transfer. (A blonde transfer is our term for an attended transfer that is converted into a blind transfer. :)) Party B could be the operator. When party B hangs up, he assumes that he is out of the original call entirely. The party B channel will be in this condition while party C is ringing, while attempting to recall party B, and while waiting between call attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to fix the problem. It will replace the party B channel technology with a NULL channel driver to complete hanging up the party B channel technology. The consequences of this code is that the 'h' extension will not be able to access any channel technology specific information like SIP statistics for the call. ATXFER_NULL_TECH is not defined by default. ******** (closes issue #17999) Reported by: iskatel Tested by: rmudgett JIRA SWP-2246 (closes issue #17096) Reported by: gelo Tested by: rmudgett JIRA SWP-1192 (closes issue #18395) Reported by: shihchuan Tested by: rmudgett (closes issue #17273) Reported by: grecco Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1047/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302178 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	9be73e35de	Merged revisions 300166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300166 \| rmudgett \| 2011-01-03 17:14:55 -0600 (Mon, 03 Jan 2011) \| 11 lines Merged revisions 300165 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r300165 \| rmudgett \| 2011-01-03 17:02:13 -0600 (Mon, 03 Jan 2011) \| 4 lines Use correct variable for atxfercallbackretries config option. * Misc formatting changes. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	cf655aa1c9	Merged revisions 299088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r299088 \| lmadsen \| 2010-12-20 10:18:26 -0600 (Mon, 20 Dec 2010) \| 13 lines Merged revisions 299087 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r299087 \| lmadsen \| 2010-12-20 10:18:03 -0600 (Mon, 20 Dec 2010) \| 5 lines Note that Park() timeout is milliseconds. (closes issue #15758) Reported by: mmurdock Tested by: mmurdock, seanbright ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	5ce016b463	Merged revisions 297952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r297952 \| twilson \| 2010-12-09 14:48:44 -0600 (Thu, 09 Dec 2010) \| 10 lines Don't crash after Set(CDR(userfield)=...) in ast_bridge_call Instead of setting peer->cdr = NULL, set it to not post. (closes issue #18415) Reported by: macbrody Patches: patch-18415 uploaded by jsolares (license 1167) Tested by: jsolares, twilson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297956 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	ce6abd6bf7	Merged revisions 289340 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289340 \| qwell \| 2010-09-29 16:12:43 -0500 (Wed, 29 Sep 2010) \| 22 lines Merged revisions 289339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289339 \| qwell \| 2010-09-29 16:03:47 -0500 (Wed, 29 Sep 2010) \| 15 lines Merged revisions 289338 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289338 \| qwell \| 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) \| 8 lines Allow a manager originate to succeed on forwarded devices. The timeout to wait for an answer was being set to 0 when a device forwarded to another extension. We don't always need the timeout set like this, so make it an optional parameter, and don't use it in this case. ABE-2544 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	e86c254b79	Merged revisions 287897 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287897 \| rmudgett \| 2010-09-21 10:53:19 -0500 (Tue, 21 Sep 2010) \| 1 line Cut-n-paste error in builtin_blindtransfer(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287898 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	eee14db850	Merged revisions 287020 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287020 \| jpeeler \| 2010-09-15 15:58:39 -0500 (Wed, 15 Sep 2010) \| 1 line fix uninintialized variable ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	41b95ee887	Merged revisions 286931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286931 \| jpeeler \| 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) \| 16 lines Add parking extension for non-default parking lots. This is a new feature that allows for parking to custom parking lots to be accessed directly, rather than with channel variables or by changing the default parking lot. The extension is set with the parkext option just as the default parking lot is done. Also, the manager action has been updated to optionally allow a specified parking lot. (closes issue #14882) Reported by: vmikhnevych Patches: patch_14882.txt uploaded by mnick (license 874) modified by me Review: https://reviewboard.asterisk.org/r/884/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	a6adb398e9	Merged revisions 286558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286558 \| tilghman \| 2010-09-13 18:50:34 -0500 (Mon, 13 Sep 2010) \| 9 lines Merged revisions 286557 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r286557 \| tilghman \| 2010-09-13 18:48:51 -0500 (Mon, 13 Sep 2010) \| 2 lines C precedence got me ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	77433168ea	Merged revisions 286528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286528 \| tilghman \| 2010-09-13 18:12:21 -0500 (Mon, 13 Sep 2010) \| 9 lines Merged revisions 286527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r286527 \| tilghman \| 2010-09-13 18:03:26 -0500 (Mon, 13 Sep 2010) \| 2 lines Refactor conversion to ast_poll() to fix callparking regression. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Olle Johansson	3335c96157	Whitespace cleanup and reformatting with { and } git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286329 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	4e0612340e	Merged revisions 285371 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285371 \| rmudgett \| 2010-09-07 16:08:35 -0500 (Tue, 07 Sep 2010) \| 1 line Fix cut-n-paste error. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285372 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	5eae9f44f7	Merged revisions 284597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284597 \| tilghman \| 2010-09-02 00:00:34 -0500 (Thu, 02 Sep 2010) \| 29 lines Merged revisions 284593,284595 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284593 \| tilghman \| 2010-09-01 17:59:50 -0500 (Wed, 01 Sep 2010) \| 18 lines Merged revisions 284478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284478 \| tilghman \| 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) \| 11 lines Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows. This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing a potential crash bug in all supported releases. (closes issue #17678) Reported by: russell Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select Review: https://reviewboard.asterisk.org/r/824/ ........ ................ r284595 \| tilghman \| 2010-09-01 22:57:43 -0500 (Wed, 01 Sep 2010) \| 2 lines Failed to rerun bootstrap.sh after last commit ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	a9e49f4e45	Update documentation for 'comebacktoorigin' in featuers.conf. The documentation for this option did not match the code. Fix that along with some minor cleanups to the code along the way. Document a slight change in behavior (to something that was previously undocumented) in UPGRADE.txt. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	ef95349d1c	Merged revisions 278167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278167 \| tilghman \| 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010) \| 4 lines Do not queue up DTMF frames while a call is on hold. (Fixes ABE-2110) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278272 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jean Galarneau	e533a48c16	Merged revisions 277906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277906 \| jeang \| 2010-07-19 15:16:36 -0500 (Mon, 19 Jul 2010) \| 7 lines Avoid trying to pickup a parked extension before the park operation is completed. A crash could occur if the extension is picked up while the parking extension is being announced. Testing pu->notquiteyet while searching for a parked extension resolves this crash. (ABE-2418) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tim Ringenbach	3442f13da4	Merged revisions 277625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277625 \| tringenbach \| 2010-07-16 17:43:39 -0500 (Fri, 16 Jul 2010) \| 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on attended transfer. ast_bridge_call() clears AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer, ast_bridge_call() is called for a second bridge on the same channel, and it clears that flag, which still needs to get set for when the original ast_bridge_call() gets control back and checks it. Review: https://reviewboard.asterisk.org/r/741 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277657 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ec37ffbdaf	ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char tag; }; struct ast_party_dialed { struct { char str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	8ae46b53a8	Merged revisions 276123 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r276123 \| russell \| 2010-07-13 14:06:53 -0500 (Tue, 13 Jul 2010) \| 2 lines Use chan->cdr instead of chan_cdr (just like peer->cdr instead of peer_cdr in the last commit). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	ea1307d9ad	Merged revisions 275994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275994 \| russell \| 2010-07-13 11:51:18 -0500 (Tue, 13 Jul 2010) \| 14 lines Access peer->cdr directly instead of through a saved off reference. At this point in the code, it is possible that peer_cdr may be invalid. Specifically, in the blind transfer code, CDRs are swapped between channels. So, peer_cdr is no longer == peer->cdr. The scenario that exposed a crash in this code was a blind transfer that hit the system call limit, causing the transferee channel to get destroyed after the transfer attempt failed. Even if it succeeds and this code doesn't crash, this code was still trying to reset a CDR on a channel that was now owned by a different thread, which is a BadThing(tm). (ABE-2417) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	b4ba8548e1	Fix some issues related to dynamic feature groups in features.conf. The bridge handling code did not properly consider feature groups when setting parameters that would affect whether or not a native bridge would be attempted. If DYNAMIC_FEATURES only include a feature group, a native bridge would occur that may prevent features from working. Fix a bug in verbose output that would show the key mapping as empty if it was using the default mapping and not a custom mapping in the feature group. Add feature groups to the output of "features show". Adjust the feature execution logic to match that of the logic when executing a feature that was not configured through a feature group. Update features.conf.sample to show that an '=' is still required if using the default key mapping from [applicationmap]. Finally, clean up a little bit of formatting to better coform to coding guidelines while in the area. (closes issue #17589) Reported by: lmadsen Patches: issue_17589.rev4.txt uploaded by russell (license 2) Tested by: russell, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	eaaeb7a1bc	Add missing ao2_iterator_destroy(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	da8450323f	Kill some startup warnings and errors and make some messages more helpful in tracking down the source. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	6d82dbb905	fixes attended transfer behavior when both transferee and transferer hung up If both the transferer and transferee of a attended transfer hangup before the new channel picks up, the new channel should be hung up as well as it has no endpoint to talk to. This mirrors the expected behavior used in 1.4. (closes issue #17444) Reported by: corruptor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	c672763af8	Fix some doxygen warnings. (closes issue #17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded by snuffy (license 35) Tested by: russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	afd4454c44	Generic Advice of Charge. Asterisk Generic AOC Representation - Generic AOC encode/decode routines. (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent generic encoded AOC data - Manager events for AOC-S, AOC-D, and AOC-E messages Asterisk App Support - app_dial AOC-S pass-through support on call setup - app_queue AOC-S pass-through support on call setup AOC Unit Tests - AOC Unit Tests for encode/decode routines - AOC Unit Test for manager event representation. SIP AOC Support - Pass-through of generic AOC-D and AOC-E messages to snom phones via the snom AOC specification. - Creation of chan_sip page3 flags for the addition of the new 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively supports AOC pass-through through the use of the new AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC Pass-through support - 'aoc_enable' chan_dahdi.conf option for independently enabling pass-through of AOC-S, AOC-D, AOC-E. - 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - DAHDI A() dial string option for requesting AOC services. example usage: ;requests AOC-S, AOC-D, and AOC-E on call setup exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: https://reviewboard.asterisk.org/r/552/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	b5d5cc565f	Enhancements to connected line and redirecting work. From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	8c41f2db82	Merged revisions 193391,258670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193391 \| mnicholson \| 2009-05-08 16:01:25 -0500 (Fri, 08 May 2009) \| 8 lines Set the proper disposition on originated calls. (closes issue #14167) Reported by: jpt Patches: call-file-missing-cdr2.diff uploaded by mnicholson (license 96) Tested by: dlotina, rmartinez, mnicholson ........ r258670 \| mnicholson \| 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) \| 11 lines Fix broken CDR behavior. This change allows a CDR record previously marked with disposition ANSWERED to be set as BUSY or NO ANSWER. Additionally this change partially reverts r235635 and does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated from ast_call(). To preserve proper CDR behavior, the AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in ast_bridge_call(). (closes issue #16797) Reported by: VarnishedOtter Tested by: mnicholson ........ (closes issue #16222) Reported by: telles Tested by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	9674766487	Fix incomplete CDR merge from r195881 Because res/res_features.c was removed and main/cdr.c added, these changes didn't make it to trunk and the 1.6.x branches git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	8b7a90a026	Yet another issue where the conversion of the application delimiter to comma caused an issue. Application arguments within the feature map could possibly contain a comma, which conflicts with the syntax of the features.conf configuration file. This patch allows the argument to be wrapped in parentheses or quoted, to allow the application arguments to be interpreted as a single configuration parameter. (closes issue #16646) Reported by: pinga-fogo Patches: 20100414__issue16646.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	8acef966ef	Merged revisions 253799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r253799 \| mnicholson \| 2010-03-22 14:50:00 -0500 (Mon, 22 Mar 2010) \| 4 lines Unconditionally copy the caller's account code to the called party. (related to issue #16331) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253800 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	3da9f8ed19	Resolve more compiler warnings on FreeBSD. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253540 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	9620ccf057	Fix ParkAndAnnounce not respecting parking options. The patch ensures that if a peer does not exist, parking settings are read from the channel. A unit test has been written to ensure proper operation for both standard parking and parking using masquerades. (closes issue #16592) Reported by: mwyres Patches: bug_16592.diff uploaded by snuffy (license 35) Review: https://reviewboard.asterisk.org/r/539/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251679 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	78fdaa6865	Add missing description of the PARKINGLOT variable in XML documentation. (closes issue #16743) Reported by: snuffy Patches: parkingdoc.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	da96c84c67	Send a manager event when the manager BridgeAction command is used. (closes issue #16769) Reported by: syspert Patches: bridgeaction.patch uploaded by syspert (license 938) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	469fed1cd3	Merged revisions 247651 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247651 \| mnicholson \| 2010-02-18 13:38:09 -0600 (Thu, 18 Feb 2010) \| 6 lines Copy the calling party's account code to the called party if they don't already have one. (closes issue #16331) Reported by: bluefox Tested by: mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	57c819fd5e	addition of dynamic parkinglots feature This feature allows for parkinglots to be created dynamically within the dialplan. Thanks to all who were involved with getting this patch written and tested! (closes issue #15135) Reported by: IgorG Patches: features.dynamic_park.v3.diff uploaded by IgorG (license 20) 2009090400_dynamicpark.diff.txt uploaded by mvanbaak (license 7) dynamic_parkinglot.diff uploaded by dvossel (license 671) Tested by: eliel, IgorG, acunningham, mvanbaak, zktech Review: https://reviewboard.asterisk.org/r/352/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	b527525ffc	Add some additional option support for non-default parking lots. The options are: parkedcallparking, parkedcallhangup, parkedcallrecording, and parkedcalltransfers. Previously these options were only available for the default parking lot. (closes issue #16641) Reported by: bluecrow76 Patches: asterisk-1.6.2.1-features.c.diff uploaded by bluecrow76 (license 270) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	7662c7fed2	Merged revisions 243390 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r243390 \| dvossel \| 2010-01-26 17:55:49 -0600 (Tue, 26 Jan 2010) \| 9 lines fixes bug with channel receiving wrong privileges after call parking (closes issue #16429) Reported by: Yasuhiro Konishi Patches: features.c.diff uploaded by Yasuhiro Konishi (license 947) Tested by: dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243391 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	568c057c4c	Extend max call limit duration from 24.8 days to 292+ million years. If the limit was set past MAX_INT upon answering, the call was immediately hung up due to overflow from the return of ast_tvdiff_ms (in ast_check_hangup). The time calculation functions ast_tvdiff_sec and ast_tvdiff_ms have been changed to return an int64_t to prevent overflow. Also the reporter suggested adding a message indicating the reason for the call hanging up. Given that the new limit is so much higher, the message (which would only really be useful in the overflow scenario) has been made a debug message only. (closes issue #16006) Reported by: viraptor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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309 Commits (e5210366e45056e56c27c902ad63a479d5ae7005)