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2009-11-04Merged revisions 227829 via svnmerge from mnicholson1-13/+62
https://origsvn.digium.com/svn/asterisk/trunk ................ r227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov 2009) | 17 lines Merged revisions 227827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov 2009) | 10 lines This patch modifies the Dial application to monitor the calling channel for hangups while playing back announcements. (closes issue #16005) Reported by: falves11 Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, falves11 Review: https://reviewboard.asterisk.org/r/407/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@227832 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02Merged revisions 226890 via svnmerge from file1-4/+27
https://origsvn.digium.com/svn/asterisk/trunk ................ r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) | 18 lines Merged revisions 226889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | 11 lines Fix a bug where the recorded privacy introduction file would not get removed if the caller hung up while the called party had not yet answered. This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction file under all scenarios. This was done to preserve the behavior that has existed for quite some time. (closes issue #14674) Reported by: ulogic Patches: bug14674.patch uploaded by jpeeler (license 325) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@226892 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-19Merged revisions 224567 via svnmerge from file1-4/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) | 12 lines Merged revisions 224565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 lines Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it. (closes issue #14763) Reported by: cupotka ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@224570 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-12Merged revisions 223832 via svnmerge from jpeeler1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009) | 15 lines Merged revisions 223804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines Ensure ringing continues for branched calls after progress is received While waiting for an answer, don't send progress for branched calls for which ringing was sent. (closes issue #15028) Reported by: fnordian ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@223834 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-09Merged revisions 223215 via svnmerge from mmichelson1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct 2009) | 9 lines Recorded merge of revisions 223213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri, 09 Oct 2009) | 3 lines Fix potential memory leak in app_dial.c ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@223241 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-24Merged revisions 208593 via svnmerge from russell1-1/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009) | 14 lines Merged revisions 208592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009) | 7 lines Do not log an ERROR if autoservice_stop() returns -1. This does not indicate an error. A return of -1 just means that the channel has been hung up. (reported in #asterisk-dev) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@208595 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-30Merged revisions 198285 via svnmerge from seanbright1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May 2009) | 15 lines Merged revisions 198251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May 2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we treat a missing one. (closes issue #15056) Reported by: p_lindheimer Patches: 05292009_bug15056.diff uploaded by seanbright (license 71) Tested by: p_lindheimer ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@198295 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-18Merged revisions 195162 via svnmerge from eliel1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r195162 | eliel | 2009-05-18 10:45:23 -0400 (Mon, 18 May 2009) | 9 lines Warn about the use of the application WaitExten() within a Macro(). Update applications documentation to warn the user about the use of the WaitExten() application within a Macro(). Recommend the use of Read() instead. (closes issue #14444) Reported by: ewieling ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@195167 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-20Merged revisions 189516 via svnmerge from twilson1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r189516 | twilson | 2009-04-20 16:29:29 -0500 (Mon, 20 Apr 2009) | 9 lines Merged revisions 189465 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189465 | twilson | 2009-04-20 16:10:27 -0500 (Mon, 20 Apr 2009) | 2 lines Update CDR appropriately when AST_CAUSE_NO_ANSWER is set ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@189535 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-20Merged revisions 189495 via svnmerge from twilson1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r189495 | twilson | 2009-04-20 16:24:34 -0500 (Mon, 20 Apr 2009) | 9 lines Merged revisions 189463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189463 | twilson | 2009-04-20 16:00:52 -0500 (Mon, 20 Apr 2009) | 2 lines Don't treat a NOANSWER like a CHANUNAVAIL ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@189534 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Merged revisions 183436 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r183436 | dvossel | 2009-03-19 15:30:39 -0500 (Thu, 19 Mar 2009) | 13 lines Merged revisions 183386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines Cleaning up a few things in detect disconnect patch Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect. issue #11583 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@183438 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Merged revisions 183172 via svnmerge from dvossel1-4/+25
https://origsvn.digium.com/svn/asterisk/trunk ................ r183172 | dvossel | 2009-03-19 11:28:33 -0500 (Thu, 19 Mar 2009) | 20 lines Merged revisions 183126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines Allow disconnect feature before a call is bridged feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c. (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) detect_disconnect.diff uploaded by dvossel (license 671) Tested by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@183198 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-12Merged revisions 181612 via svnmerge from file1-2/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r181612 | file | 2009-03-12 10:24:12 -0300 (Thu, 12 Mar 2009) | 5 lines Fix crash when sleep and retries argument was not given to RetryDial application. (closes issue #14647) Reported by: sherpya ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@181614 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-04Merged revisions 180120 via svnmerge from file1-2/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r180120 | file | 2009-03-04 10:39:28 -0400 (Wed, 04 Mar 2009) | 7 lines Remove duplicate 'k' and 'K' Dial options. (closes issue #14601) Reported by: alecdavis Patches: app_dial.optionk.diff.txt uploaded by alecdavis (license 585) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@180122 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-11Merged revisions 174945 via svnmerge from mmichelson1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb 2009) | 29 lines Fix 'd' option for app_dial and add new option to Answer application The 'd' option would not work for channel types which use RTP to transport DTMF digits. The only way to allow for this to work was to answer the channel if we saw that this option was enabled. I realized that this may cause issues with CDRs, specifically with giving false dispositions and answer times. I therefore modified ast_answer to take another parameter which would tell if the CDR should be marked answered. I also extended this to the Answer application so that the channel may be answered but not CDRified if desired. I also modified app_dictate and app_waitforsilence to only answer the channel if it is not already up, to help not allow for faulty CDR answer times. All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the changes except for the change to the Answer application will go in since we do not introduce new features into stable branches (closes issue #14164) Reported by: DennisD Patches: 14164.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/145 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@174947 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-31Merged revisions 172580 via svnmerge from twilson1-63/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r172580 | twilson | 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009) | 44 lines Merged revisions 172517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@172636 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-23Merged revisions 170569 via svnmerge from file1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r170569 | file | 2009-01-23 15:09:18 -0400 (Fri, 23 Jan 2009) | 11 lines Merged revisions 170568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4 lines When a call is forwarded stop any active indications. The new channel will provide an indication, if need be, itself. (closes issue #14310) Reported by: RadicAlish ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@170571 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-09Merged revisions 167935 via svnmerge from twilson1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r167935 | twilson | 2009-01-08 18:13:12 -0600 (Thu, 08 Jan 2009) | 2 lines Set peer context and exten values so MACRO_EXTEN and MACRO_CONTEXT will be set ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@167972 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-29Merged revisions 166861 via svnmerge from mmichelson1-2/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r166861 | mmichelson | 2008-12-29 12:04:52 -0600 (Mon, 29 Dec 2008) | 14 lines Update app_queue to deal with the removal of AST_PBX_KEEPALIVE When placing a call to a queue which ran a gosub on the member's channel, Asterisk would crash every time, stemming from the fact that the member's channel was being hung up unexpectedly when the Gosub completed. The necessary change was pretty much copied and pasted from app_dial's similar changes made last week. I also took the opportunity to change a LOG_DEBUG message in app_dial to use ast_debug. I am guessing this was due to a direct merge from 1.4 that was not corrected to use trunk's preferred syntax. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@166863 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-24Merged revisions 166665 via svnmerge from murf1-39/+27
https://origsvn.digium.com/svn/asterisk/trunk This merged from trunk with no conflicts. I tested mostly the 'tired' cases, and for the most part ignored the tests for reconnecting and dialing in to fetch a parked call, after the first case. ................ r166665 | murf | 2008-12-23 11:13:49 -0700 (Tue, 23 Dec 2008) | 153 lines Merged revisions 166093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In order to merge this 1.4 patch into trunk, I had to resolve some conflicts and wait for Russell to make some changes to res_agi. I re-ran all the tests; 39 calls in all, and made fairly careful notes and comparisons: I don't want this to blow up some aspect of asterisk; I completely removed the KEEPALIVE from the pbx.h decls. The first 3 scenarios involving feature park; feature xfer to 700; hookflash park to Park() app call all behave the same, don't appear to leave hung channels, and no crashes. ........ r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines This merges the masqpark branch into 1.4 These changes eliminate the need for (and use of) the KEEPALIVE return code in res_features.c; There are other places that use this result code for similar purposes at a higher level, these appear to be left alone in 1.4, but attacked in trunk. The reason these changes are being made in 1.4, is that parking ends a channel's life, in some situations, and the code in the bridge (and some other places), was not checking the result code properly, and dereferencing the channel pointer, which could lead to memory corruption and crashes. Calling the masq_park function eliminates this danger in higher levels. A series of previous commits have replaced some parking calls with masq_park, but this patch puts them ALL to rest, (except one, purposely left alone because a masquerade is done anyway), and gets rid of the code that tests the KEEPALIVE result, and the NOHANGUP_PEER result codes. While bug 13820 inspired this work, this patch does not solve all the problems mentioned there. I have tested this patch (again) to make sure I have not introduced regressions. Crashes that occurred when a parked party hung up while the parking party was listening to the numbers of the parking stall being assigned, is eliminated. These are the cases where parking code may be activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via manager. The interesting testing cases for parking are: I. A calls B, A parks B a. B hangs up while A is getting the numbers announced. b. B hangs up after A gets the announcement, but before the parking time expires c. B waits, time expires, A is redialed, A answers, B and A are connected, after which, B hangs up. d. C picks up B while still in parking lot. II. A calls B, B parks A a. A hangs up while B is getting the numbers announced. b. A hangs up after B gets the announcement, but before the parking time expires c. A waits, time expires, B is redialed, B answers, A and B are connected, after which, A hangs up. d. C picks up A while still in parking lot. Testing this throroughly involves acting all the permutations of I and II, in situations 1,2,3, and 4. Since I added a few more changes (ALL references to KEEPALIVE in the bridge code eliimated (I missed one earlier), I retested most of the above cases, and no crashes. H-extension weirdness. Current h-extension execution is not completely correct for several of the cases. For the case where A calls B, and A parks B, the 'h' exten is run on A's channel as soon as the park is accomplished. This is expected behavior. But when A calls B, and B parks A, this will be current behavior: After B parks A, B is hung up by the system, and the 'h' (hangup) exten gets run, but the channel mentioned will be a derivative of A's... Thus, if A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be those relating to Channel A. And, in the case where A is reconnected to B after the park time expires, when both parties hang up after the joyful reunion, no h-exten will be run at all. In the case where C picks up A from the parking lot, when either A or C hang up, the h-exten will be run for the C channel. CDR's are a separate issue, and not addressed here. As to WHY this strange behavior occurs, the answer lies in the procedure followed to accomplish handing over the channel to the parking manager thread. This procedure is called masquerading. In the process, a duplicate copy of the channel is created, and most of the active data is given to the new copy. The original channel gets its name changed to XXX<ZOMBIE> and keeps the PBX information for the sake of the original thread (preserving its role as a call originator, if it had this role to begin with), while the new channel is without this info and becomes a call target (a "peer"). In this case, the parking lot manager thread is handed the new (masqueraded) channel. It will not run an h-exten on the channel if it hangs up while in the parking lot. The h exten will be run on the original channel instead, in the original thread, after the bridge completes. See bug 13820 for our intentions as to how to clean up the h exten behavior. Review: http://reviewboard.digium.com/r/29/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@166730 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-18Merged revisions 165723 via svnmerge from russell1-2/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r165723 | russell | 2008-12-18 13:33:42 -0600 (Thu, 18 Dec 2008) | 14 lines Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial. This is part of an effort to completely remove AST_PBX_KEEPALIVE and other similar return codes from the source. While this usage was perfectly safe, there are others that are problematic. Since we know ahead of time that we do not want to PBX to destroy the channel, the PBX API has been changed so that information can be provided as an argument, instead, thus removing the need for the KEEPALIVE return value. Further changes to get rid of KEEPALIVE and related code is being done by murf. There is a patch up for that on review 29. Review: http://reviewboard.digium.com/r/98/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@165728 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-03Merged revisions 160626 via svnmerge from mmichelson1-3/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r160626 | mmichelson | 2008-12-03 12:37:46 -0600 (Wed, 03 Dec 2008) | 16 lines Add some safety measures when using gosub, especially when using the options for app_dial and app_queue to run a gosub when the call is answered. * Check for the existence of the gosub target in gosub_exec. If it is nonexistent, then this will cause errors when we attempt to actually run the gosub, including a definite memory leak and potential crashes. Return an error in this situation * Check the return value of pbx_exec in app_dial and app_queue before attempting to actually run the gosub routine. If there was an error, we should not attempt to run the gosub. * Change a '|' to a ',' in app_queue. * Add some extra curly braces where they had been missing previously. (closes issue #13548) Reported by: fiddur ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@160628 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-26Merged revisions 159554 via svnmerge from mmichelson1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r159554 | mmichelson | 2008-11-26 13:57:11 -0600 (Wed, 26 Nov 2008) | 19 lines Add some necessary hangup commands in the case that forwarding a call fails 1) Hang up the original destination if the local channel cannot be requested. 2) Hang up the local channel (in addition to the original destination) if ast_call fails when calling the newly created local channel. This prevents channels from sticking around forever in the case of a botched call forward (e.g. to an extension which does not exist). (closes issue #13764) Reported by: davidw Patches: 13764_v2.patch uploaded by putnopvut (license 60) Tested by: putnopvut, davidw ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@159561 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-20Merged revisions 158066 via svnmerge from mmichelson1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r158066 | mmichelson | 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines Merged revisions 158053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines Make sure to set the hangup cause on the calling channel in the case that ast_call() fails. For incoming SIP channels, this was causing us to send a 603 instead of a 486 when the call-limit was reached on the destination channel. (closes issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded by putnopvut (license 60) Tested by: blitzrage ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@158067 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-18Merged revisions 157306 via svnmerge from mmichelson1-1/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r157306 | mmichelson | 2008-11-18 12:31:08 -0600 (Tue, 18 Nov 2008) | 20 lines Merged revisions 157305 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines Fix a crash in the end_bridge_callback of app_dial and app_followme which would occur at the end of an attended transfer. The error occurred because we initially stored a pointer to an ast_channel which then was hung up due to a masquerade. This commit adds a "fixup" callback to the bridge_config structure to allow for end_bridge_callback_data to be changed in the case that a new channel pointer is needed for the end_bridge_callback. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@157308 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-17Merged revisions 157253 via svnmerge from tilghman1-3/+12
https://origsvn.digium.com/svn/asterisk/trunk ........ r157253 | tilghman | 2008-11-17 16:25:06 -0600 (Mon, 17 Nov 2008) | 8 lines Can't use items duplicated off the stack frame in an element returned from a function: in these cases, we have to use the heap, or garbage will result. (closes issue #13898) Reported by: alecdavis Patches: 20081114__bug13898__2.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@157255 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-12Merged revisions 156388 via svnmerge from tilghman1-12/+15
https://origsvn.digium.com/svn/asterisk/trunk ................ r156388 | tilghman | 2008-11-12 15:34:51 -0600 (Wed, 12 Nov 2008) | 12 lines Merged revisions 156386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008) | 5 lines When using call limits under 1 second, infinite call lengths are allowed, instead. (closes issue #13851) Reported by: ruddy ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@156389 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-12Merged revisions 156169 via svnmerge from mmichelson1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r156169 | mmichelson | 2008-11-12 11:41:56 -0600 (Wed, 12 Nov 2008) | 15 lines Merged revisions 156167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov 2008) | 7 lines When doing some tests, I was having a crash at the end of every call if an attended transfer occurred during the call. I traced the cause to the CDR on one of the channels being NULL. murf suggested a check in the end bridge callback to be sure the CDR is non-NULL before proceeding, so that's what I'm adding. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@156171 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-09Merged revisions 155554 via svnmerge from seanbright1-21/+22
https://origsvn.digium.com/svn/asterisk/trunk ................ r155554 | seanbright | 2008-11-08 20:27:00 -0500 (Sat, 08 Nov 2008) | 14 lines Merged revisions 155553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov 2008) | 6 lines Use static functions here instead of nested ones. This requires a small change to the ast_bridge_config struct as well. To understand the reason for this change, see the following post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@155556 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-02import gcc 4.3.2 warning fixes from trunk, with a few changes specific to ↵kpfleming1-2/+8
this branch git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@153710 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-31Merged revisions 153181 via svnmerge from twilson1-12/+25
https://origsvn.digium.com/svn/asterisk/trunk ........ r153181 | twilson | 2008-10-31 13:55:33 -0500 (Fri, 31 Oct 2008) | 5 lines Recent CDR fixes moved execution of the 'h' exten into the bridging code, so variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call (closes issue #13793) Reported by: greenfieldtech ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@153266 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-29Merged revisions 152605 via svnmerge from murf1-10/+20
https://origsvn.digium.com/svn/asterisk/trunk ................ r152605 | murf | 2008-10-28 23:47:13 -0600 (Tue, 28 Oct 2008) | 22 lines Merged revisions 152538 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) | 14 lines A little documentation cross-ref between features and dial and queue... I wasted some time (stupidly) trying to get the one-touch parking stuff working, because it didn't occur to me that I had to also have the corresponding options in the dial command! Duh! (In all this time, I never set this up before!) So, to keep some poor fool from suffering the same fate, I made the features.conf.sample file mention the corresponding opts in dial/queue; and the docs for dial/app specifically mention the corresponding decls in the feature.conf file. I hope this doesn't spoil some vast, eternal plan... ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@152606 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-29Merged revisions 152536 via svnmerge from murf1-42/+51
https://origsvn.digium.com/svn/asterisk/trunk ................ r152536 | murf | 2008-10-28 23:01:00 -0600 (Tue, 28 Oct 2008) | 57 lines Merged revisions 152535 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines The magic trick to avoid this crash is not to try to find the channel by name in the list, which is slow and resource consuming, but rather to pay attention to the result codes from the ast_bridge_call, to which I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are returned when a channel is parked. Why? because CDR's aren't generated via parking, so nothing is needed, but if a transfer occurred, there are critical things I need. If you get AST_PBX_KEEPALIVE, then don't touch the channel pointer. If you get AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then don't touch the peer pointer. Updated the several places where the results from a bridge were not being properly obeyed, and fixed some code I had introduced so that the results of the bridge were not overridden (in trunk). All the places that previously tested for AST_PBX_NO_HANGUP_PEER now have to check for both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common parking scenarios: 1. A calls B; B answers; A parks B; B hangs up while A is getting the parking slot announcement, immediately after being put on hold. 2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but before the park times out. 3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold. 4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out. No crash. I also ran the scenarios above against valgrind, and accesses looked good. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@152537 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-28Merged revisions 152369 via svnmerge from tilghman1-0/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r152369 | tilghman | 2008-10-28 12:07:39 -0500 (Tue, 28 Oct 2008) | 15 lines Merged revisions 152368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008) | 8 lines Reset all DIAL variables back to blank, in case Dial is called multiple times per call (which could otherwise lead to inconsistent status reports). (closes issue #13216) Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded by Corydon76 (license 14) Tested by: ruddy ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@152370 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-14Merged revisions 149279 via svnmerge from mmichelson1-2/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r149279 | mmichelson | 2008-10-14 18:57:46 -0500 (Tue, 14 Oct 2008) | 7 lines When specifying an invalid timeout to Dial, take it to mean that no timeout is desired. (closes issue #13625) Reported by: atis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@149280 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-07Merged revisions 147050 via svnmerge from seanbright1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r147050 | seanbright | 2008-10-07 08:01:36 -0400 (Tue, 07 Oct 2008) | 8 lines Make sure to compare the correct number of characters when special-casing our DAHDI operator mode stuff. Technically, it would work fine, as 'DAH' is currently unique amongst our channel technologies, but as Jared points out: <@jsmith> Sure... as long as the technology starts whith DAH.... but it could be DAHDOO! ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@147052 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-13Merged revisions 143031 via svnmerge from tilghman1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r143031 | tilghman | 2008-09-13 08:54:15 -0500 (Sat, 13 Sep 2008) | 8 lines Repair IAXVAR implementation so that it works again (regression?) (closes issue #13354) Reported by: adomjan Patches: 20080828__bug13354.diff.txt uploaded by Corydon76 (license 14) 20080829__bug13354__1.6.0.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, adomjan ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@143033 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Merged revisions 142676 via svnmerge from murf1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r142676 | murf | 2008-09-11 22:50:48 -0600 (Thu, 11 Sep 2008) | 40 lines Merged revisions 142675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | 29 lines Tested by: sergee, murf, chris-mac, andrew, KNK This is a "second attempt" to restore the previous "endbeforeh" behavior in 1.4 and up. In order to capture information concerning all the legs of transfers in all their infinite combinations, I was forced to this particular solution by a chain of logical necessities, the first being that I was not allowed to rewrite the CDR mechanism from the ground up! This change basically leaves the original machinery alone, which allows IVR and local channel type situations to generate CDR's as normal, but a channel flag can be set to suppress the normal running of the h exten. That flag would be set by the code that runs the h exten from the ast_bridge_call routine, to prevent the h exten from being run twice. Also, a flag in the ast_bridge_config struct passed into ast_bridge_call can be used to suppress the running of the h exten in that routine. This would happen, for instance, if you use the 'g' option in the Dial app. Running this routine 'early' allows not only the CDR() func to be used in the h extension for reading CDR variables, but also allows them to be modified before the CDR is posted to the backends. While I dearly hope that this patch overcomes all problems, and introduces no new problems, reality suggests that surely someone will have problems. In this case, please re-open 13251 (or 13289), and we'll see if we can't fix any remaining issues. ** trunk note: some code to suppress the h exten being run from app_queue was added; for the 'continue' option available only in trunk/1.6.x. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@142678 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-22Merged revisions 139627 via svnmerge from murf1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r139627 | murf | 2008-08-22 16:03:13 -0600 (Fri, 22 Aug 2008) | 59 lines Merged revisions 139347 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) | 47 lines (closes issue #13251) Reported by: sergee Tested by: murf THis is a bold move for a static release fix, but I wouldn't have made it if I didn't feel confident (at least a *bit* confident) that it wouldn't mess everyone up. The reasoning goes something like this: 1. We simply cannot do anything with CDR's at the current point (in pbx.c, after the __ast_pbx_run loop). It's way too late to have any affect on the CDRs. The CDR is already posted and gone, and the remnants have been cleared. 2. I was very much afraid that moving the running of the 'h' extension down into the bridge code (where it would be now practical to do it), would result in a lot more calls to the 'h' exten, so I implemented it as another exten under another name, but found, to my pleasant surprise, that there was a 1:1 correspondence to the running of the 'h' exten in the pbx_run loop, and the new spot at the end of the bridge. So, I ifdef'd out the current 'h' loop, and moved it into the bridge code. The only difference I can see is the stuff about the AST_PBX_KEEPALIVE, and hopefully, if this is still an important decision point, I can replicate it if there are complaints. To be perfectly honest, the KEEPALIVE situation is not totally clear to me, and how it relates to a post-bridge situation is less clear. I suspect the users will point out everything in total clarity if this steps on anyone's toes! 3. I temporarily swap the bridge_cdr into the channel before running the 'h' exten, which makes it possible for users to edit the cdr before it goes out the door. And, of course, with the endbeforehexten config var set, the users can also get at the billsec/duration vals. After the h exten finishes, the cdr is swapped back and processing continues as normal. Please, all who deal with CDR's, please test this version of Asterisk, and file bug reports as appropriate! ........ I also made a little fix to the app_dial's 'e' option, that is related to my updates. ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@139629 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-10More RSW merges. Everything from apps/ except for the big offendersseanbright1-9/+9
app_voicemail and app_queue. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@137055 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-05Merged revisions 135799 via svnmerge from murf1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | 34 lines (closes issue #12982) Reported by: bcnit Tested by: murf I discovered that also, in the previous bug fixes and changes, the cdr.conf 'unanswered' option is not being obeyed, so I fixed this. And, yes, there are two 'answer' times involved in this scenario, and I would agree with you, that the first answer time is the time that should appear in the CDR. (the second 'answer' time is the time that the bridge was begun). I made the necessary adjustments, recording the first answer time into the peer cdr, and then using that to override the bridge cdr's value. To get the 'unanswered' CDRs to appear, I purposely output them, using the dial cmd to mark them as DIALED (with a new flag), and outputting them if they bear that flag, and you are in the right mode. I also corrected one small mention of the Zap device to equally consider the dahdi device. I heavily tested 10-sec-wait macros in dial, and without the macro call; I tested hangups while the macro was running vs. letting the macro complete and the bridge form. Looks OK. Removed all the instrumentation and debug. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135821 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-05make datastore creation and destruction a generic API since it is not really ↵kpfleming1-4/+4
channel related, and add the ability to add/find/remove datastores to manager sessions git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135680 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-14Merged revisions 130792 via svnmerge from mmichelson1-6/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul 2008) | 8 lines Add a check to the CAN_EARLY_BRIDGE macro in app_dial to be sure there are no audiohooks present on the channels involved. This fixed a one-way audio situation I had in my test setup. I couldn't find any open issues that suggested one-way audio with regards to mixmonitor (or other audiohook) usage, though. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130794 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-08Merged revisions 129149 via svnmerge from tilghman1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) | 8 lines Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not registered. (closes issue #12885) Reported by: ibc Patches: 20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14) Tested by: ibc ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129152 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-01another minor ast_channel memory size decrease... for nearly all channels, ↵kpfleming1-6/+3
'dialcontext' is only going to be set once during the channel's lifetime, so make it a string field instead of a char array git-svn-id: http://svn.digium.com/svn/asterisk/trunk@126960 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-26Improve consistency between app_dial and app_queue with regardsmmichelson1-2/+0
to how language is handled between two channels whose native language is different. Prior to this patch, app_dial would have the callee inherit the caller's language, and app_queue would not. After this patch, app_dial no longer has the language inheritance capability. This seems to make the most sense since it seems more natural for a person to hear files played back in his/her native language instead of the language of the person on the far end of the call. See the CHANGES file for hints on how to keep the previous behavior of app_dial if desired. (closes issue #12489) Reported by: bcnit git-svn-id: http://svn.digium.com/svn/asterisk/trunk@125647 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-18Channel lock janitor -- add locks around retrieval of channel variablestilghman1-13/+43
(closes issue #12840) Reported by: pputman Patches: app_dial_threadsafe3.patch uploaded by pputman (license 81) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@123648 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-16(closes issue #12689)murf1-1/+1
Reported by: ys Many thanks to ys for doing the research on this problem. I didn't think it would be best to unlock the contexts and then relock them after the remove_extension2() call, so I added an extra arg to remove_extension2() and set it appropriately in each call. There were not that many. I considered forcing the code to lock the contexts before the call to remove_extension2(), but that would require a slightly greater degree of changes, especially since the find_context_locked is local to pbx.c I did a simple sanity test to make sure the code doesn't mess things up in general. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@123165 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-12Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. ↵jpeeler1-5/+5
Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122234 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-02Merged revisions 119530 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119530 | russell | 2008-06-01 20:03:22 -0500 (Sun, 01 Jun 2008) | 2 lines Fix another typo in documentation ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119531 f38db490-d61c-443f-a65b-d21fe96a405b
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