394 Commits (f55f994661d4b4a08b58a0df01b693acc33bfec2)

Author	SHA1	Message	Date
Terry Wilson	a671c82ec2	Merged revisions 167935 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@167972 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	45ac5e6838	Merged revisions 166861 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166863 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	a7aeaf341b	Merged revisions 166665 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	52f5e8fcfb	Merged revisions 165723 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@165728 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	ec71774eae	Merged revisions 160626 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@160628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	81826df05d	Merged revisions 159554 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@159561 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	7122ee6adb	Merged revisions 158066 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@158067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	892f98a1e6	Merged revisions 157306 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@157308 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ed1113b08e	Merged revisions 157253 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@157255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	1b8217193c	Merged revisions 156388 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@156389 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	0c0fdebd9f	Merged revisions 156169 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@156171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	839cb83ea3	Merged revisions 155554 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@155556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	1036849a42	import gcc 4.3.2 warning fixes from trunk, with a few changes specific to this branch ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@153710 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	239f1d53bc	Merged revisions 153181 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@153266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	039de710a7	Merged revisions 152605 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@152606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	f1da1c3957	Merged revisions 152536 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@152537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	a7a57a4f61	Merged revisions 152369 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@152370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	28aafe781b	Merged revisions 149279 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@149280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	b0d673495e	Merged revisions 147050 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@147052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	25dd2fa2f3	Merged revisions 143031 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@143033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	bc73329607	Merged revisions 142676 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@142678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	41138ad333	Merged revisions 139627 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@139629 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	3ffb39833b	More RSW merges. Everything from apps/ except for the big offenders ... app_voicemail and app_queue. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137055 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	5eaf8450d6	Merged revisions 135799 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@135821 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	7df8b8b848	make datastore creation and destruction a generic API since it is not really channel related, and add the ability to add/find/remove datastores to manager sessions ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	bd1bb0d0e2	Merged revisions 130792 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@130794 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	da03cdd174	Merged revisions 129149 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@129152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	da14954bdc	another minor ast_channel memory size decrease... for nearly all channels, '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: https://origsvn.digium.com/svn/asterisk/trunk@126960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	0178d0ccd6	Improve consistency between app_dial and app_queue with regards ... 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: https://origsvn.digium.com/svn/asterisk/trunk@125647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	90867b2b0c	Channel lock janitor -- add locks around retrieval of channel variables ... (closes issue #12840) Reported by: pputman Patches: app_dial_threadsafe3.patch uploaded by pputman (license 81) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	f4c85ebd22	(closes issue #12689) ... 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: https://origsvn.digium.com/svn/asterisk/trunk@123165 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	ef3b214728	Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	db5f489865	Merged revisions 119530 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@119531 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	0da2734cb5	Merged revisions 119478 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119478 | mvanbaak | 2008-06-01 22:47:55 +0200 (Sun, 01 Jun 2008) | 2 lines small typo fix 'retires' => 'retries' ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	c7191467d2	Add native AGI command GOSUB, as invoking Gosub with EXEC does not work ... properly. (closes issue #12760) Reported by: Corydon76 Patches: 20080530__bug12760.diff.txt uploaded by Corydon76 (license 14) Tested by: tim_ringenbach, Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119296 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	f1e9371da8	- revert change to ast_queue_hangup and create ast_queue_hangup_with_cause ... - make data member of the ast_frame struct a named union instead of a void Recently the ast_queue_hangup function got a new parameter, the hangupcause Feedback came in that this is no good and that instead a new function should be created. This I did. The hangupcause was stored in the seqno member of the ast_frame struct. This is not very elegant, and since there's already a data member that one should be used. Problem is, this member was a void *. Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone wants to store another type in there in the future. This commit is so massive, because all ast_frame.data uses have to be altered to ast_frame.data.data Thanks russellb and kpfleming for the feedback. (closes issue #12674) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	b5a127daac	Modify TIMEOUT() to be accurate down to the millisecond. ... (closes issue #10540) Reported by: spendergrass Patches: 20080417__bug10540.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115076 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	6a81da594d	Add incomplete matching to PBX code and app_dial ... (closes issue #12351) Reported by: Corydon76 Patches: 20080402__pbx_incomplete__3.diff.txt uploaded by Corydon76 (license 14) pbx_incomplete_with_timeout.diff uploaded by fabled (license 448) Tested by: Corydon76, fabled git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114773 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	463a5dbd0a	Whitespace changes only ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114667 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	08e674bce0	Pass the hangup cause all the way to the calling app/channel. ... (closes issue #11328) Reported by: rain Patches: 20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14) brought up-to-date to trunk by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	df7cb6b30b	Merged revisions 114112 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114112 | mmichelson | 2008-04-14 11:24:22 -0500 (Mon, 14 Apr 2008) | 9 lines If the datastore has been moved to another channel due to a masquerade, then freeing the datastore here causes an eventual double free when the new channel hangs up. We should only free the datastore if we were able to successfully remove it from the channel we are referencing (i.e. the datastore was not moved). (closes issue #12359) Reported by: pguido ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	1c691646a9	Permit callee to continue in the dialplan, after caller has hung up. ... (closes issue #11954) Reported by: johan Patches: app_dial_rev104031.patch uploaded by johan (license 334) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113649 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	5176911dfe	Remove some redundant logic from wait_for_answer. This also let's us get rid of one of ... those XXX comments from the code. The redundancy occurs because the 'single' flag implies that the 'r' and 'm' flags are not set, so there's no need to explicitly check them again. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	af7e1964f2	Merged revisions 107016 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107016 | file | 2008-03-10 11:33:02 -0300 (Mon, 10 Mar 2008) | 7 lines Move where unanswered CDRs are dropped to the CDR core, not everything uses app_dial. (closes issue #11516) Reported by: ys Patches: branch_1.4_cdr.diff uploaded by ys (license 281) Tested by: anest, jcapp, dartvader ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107017 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	377e51c4d4	(closes issue #6002) ... Reported by: rizzo Tested by: murf Proposal of the changes to be made, and then an announcement of how they were accomplished: http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html and: http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html Here is a recap, file by file, of what I have done: pbx/pbx_config.c pbx/pbx_ael.c All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set. Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it is just as necessary to have the TABLE available. This is because the list/table in question might not be the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global position when things are ready. We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing "find" and "create", as all existing usages used both in tandem anyway. pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and then call merge_contexts_and_delete, which will merge (now) existing contexts and priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will lock down the contexts, swap the lists and tables, and unlock (real quick), and then destroy the old dialplan. chan_sip.c chan_iax.c chan_skinny.c All the channel drivers that would add regcontexts now use the ast_context_find_or_create now. chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered. apps/app_meetme.c apps/app_dial.c apps/app_queue.c All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead. include/asterisk/pbx.h ast_context_create() is removed. Find_or_create_ is the new method. ast_context_find_or_create() interface gets the hashtab added. ast_merge_contexts_and_delete() gets the local hashtab arg added. ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking. ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael ast_hashtab_hash_contexts was in like fashion make public. include/asterisk/pval.h ast_compile_ael2() interface changed to include the local hashtab table ptr. main/features.c For the sake of the parking context, we use ast_context_find_or_create(). main/pbx.c I changed all the "tree" names to "table" instead. That's because the original implementation was based on binary trees. (had a free library). Then I moved to hashtabs. Now, the names move forward too. refcount field added to contexts, so you can keep track of how many modules wanted this context to exist. Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING. Added some calls to ast_verb(3,...) for debug messages Lots of little mods to ast_context_remove_extension2, which is now excersized in ways it was not previously; one definite bug fixed. find_or_create was upgraded to handle both local lists/tables as well as the globals. context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables ast_merge_contexts_and_delete() was heavily modified. ast_add_extension2() was also upgraded to handle changes. the context_destroy() code was re-engineered to handle the new way of doing things, by exten/prio instead of by context. res/ael/pval.c res/ael/ael.tab.c res/ael/ael.tab.h res/ael/ael.y res/ael/ael_lex.c res/ael/ael.flex utils/ael_main.c utils/extconf.c utils/conf2ael.c utils/Makefile Had to change the interface to ast_compile_ael2(), to include the hashtab ptr. This ended up involving several external apps. The main gotcha was I had to include lock.h and hashtab.h in several places. As a side note, I tested this stuff pretty thoroughly, I replicated the problems originally reported by Luigi, and made triply sure that reloads worked, and everything worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into trunk, that did not appear in my tests of bug6002. How's this for verbose commit messages? git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	496adc6fc0	Merged revisions 106235 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things. (closes issue #12148) Reported by: jcomellas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106239 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	8a411ccf83	Create a centralized configuration option for silencethreshold ... (closes issue #11236) Reported by: philipps Patches: 20080218__bug11236.diff.txt uploaded by Corydon76 (license 14) Tested by: philipps git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	7d1891d5c3	Asterisk, when parking can drop rights a caller when a parking timeout occurs. Also, when doing built-in attended transfers, sometimes incorrectly passes rights from the transferrer to the transferee. This patch tries to fixes the parking issue and lays some groundwork for later fixing the transfer issue. ... (closes issue #11520) Reported by: pliew Tested by: otherwiseguy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105477 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	4dccb58fb7	whitespace fixes only. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	fe9821cc10	Get rid of any remaining ast_verbose calls in the code in favor of ... ast_verb (closes issue #11934) Reported by: mvanbaak Patches: 20080205_astverb-2.diff.txt uploaded by mvanbaak (license 7) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@102525 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago