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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
2011-02-07Merged revisions 306674 via svnmerge from Terry Wilson
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
2011-02-04Replace ast_log(LOG_DEBUG, ...) with ast_debug()Paul Belanger
(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
2011-02-03Merged revisions 306124 via svnmerge from Jeff Peeler
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
2011-02-03Asterisk media architecture conversion - no more format bitfieldsDavid Vossel
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
2011-01-26Merged revisions 304339 via svnmerge from Jeff Peeler
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
2011-01-25Merged revisions 304007 via svnmerge from Richard Mudgett
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
2011-01-24Merged revisions 303549 via svnmerge from Russell Bryant
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
2011-01-20Merged revisions 303107 via svnmerge from Shaun Ruffell
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
2011-01-19Merged revisions 302713 via svnmerge from Richard Mudgett
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
2011-01-19Merged revisions 302552 via svnmerge from Sean Bright
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
2011-01-18Merged revisions 302318 via svnmerge from Richard Mudgett
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
2011-01-18Merged revisions 302174 via svnmerge from Richard Mudgett
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
2011-01-03Merged revisions 300166 via svnmerge from Richard Mudgett
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
2010-12-20Merged revisions 299088 via svnmerge from Leif Madsen
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
2010-12-09Merged revisions 297952 via svnmerge from Terry Wilson
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
2010-09-29Merged revisions 289340 via svnmerge from Jason Parker
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
2010-09-21Merged revisions 287897 via svnmerge from Richard Mudgett
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
2010-09-15Merged revisions 287020 via svnmerge from Jeff Peeler
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
2010-09-15Merged revisions 286931 via svnmerge from Jeff Peeler
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
2010-09-13Merged revisions 286558 via svnmerge from Tilghman Lesher
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
2010-09-13Merged revisions 286528 via svnmerge from Tilghman Lesher
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
2010-09-11Whitespace cleanup and reformatting with { and }Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-07Merged revisions 285371 via svnmerge from Richard Mudgett
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
2010-09-02Merged revisions 284597 via svnmerge from Tilghman Lesher
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
2010-07-21Update documentation for 'comebacktoorigin' in featuers.conf.Russell Bryant
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
2010-07-20Merged revisions 278167 via svnmerge from Tilghman Lesher
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
2010-07-19Merged revisions 277906 via svnmerge from Jean Galarneau
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
2010-07-16Merged revisions 277625 via svnmerge from Tim Ringenbach
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
2010-07-14ast_callerid restructuringRichard Mudgett
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
2010-07-13Merged revisions 276123 via svnmerge from Russell Bryant
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
2010-07-13Merged revisions 275994 via svnmerge from Russell Bryant
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
2010-07-09Fix some issues related to dynamic feature groups in features.conf.Russell Bryant
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
2010-07-09Add missing ao2_iterator_destroy().Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275310 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-09Kill some startup warnings and errors and make some messages more helpful in ↵Tilghman Lesher
tracking down the source. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-22fixes attended transfer behavior when both transferee and transferer hung upDavid Vossel
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
2010-06-08Fix some doxygen warnings.Leif Madsen
(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
2010-06-02Generic Advice of Charge.Richard Mudgett
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
2010-05-17Enhancements to connected line and redirecting work.Mark Michelson
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
2010-04-22Merged revisions 193391,258670 via svnmerge from Matthew Nicholson
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
2010-04-19Fix incomplete CDR merge from r195881Terry Wilson
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
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