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2009-02-05Merged revisions 173776 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r173776 | mmichelson | 2009-02-05 17:48:48 -0600 (Thu, 05 Feb 2009) | 14 lines Update extensions.conf.sample to be correct. In trunk, the only necessary change pointed out was that the call to ChanIsAvail uses an option that has been removed. For the 1.6.1 branch, however, it appears that the sample file is badly in need of updating since there are |'s used all over the place there. My tentative plan is just to copy trunk's sample config file to those branches since the info there is most up-to-date and should be correct for use in 1.6.1 Thanks to macli in #asterisk-dev for bringing this up ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173777 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05Merged revisions 173773 via svnmerge from mmichelson1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r173773 | mmichelson | 2009-02-05 17:28:19 -0600 (Thu, 05 Feb 2009) | 7 lines Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage (closes issue #13905) Reported by: jaroth Patches: foldermove_v2.patch uploaded by jaroth (license 50) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173774 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05Merged revisions 173697 via svnmerge from jpeeler1-2/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r173697 | jpeeler | 2009-02-05 15:00:26 -0600 (Thu, 05 Feb 2009) | 18 lines Merged revisions 173696 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) | 12 lines Add new configuration option to make shared IMAP mailboxes function as expected. The new option is "imapvmshareid" which is an ID to tag multiple mailboxes using the same IMAP storage location to function as one mailbox. This allows all messages to be retrieved for any user in the group. The patch alters the 'X-Asterisk-VM-Extension' header that is responsible for matching voicemails for a given user. (closes issue #13673) Reported by: howardwilkinson ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173698 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05Merged revisions 173693 via svnmerge from mmichelson1-6/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r173693 | mmichelson | 2009-02-05 14:30:45 -0600 (Thu, 05 Feb 2009) | 20 lines Merged revisions 173692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) | 12 lines Fix situations where queue members could be autopaused unexpectedly Specifically, this patch prevents us from autopausing members when we receive a busy or congestion frame from them. (closes issue #14376) Reported by: fiddur Patches: 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173694 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05Merged revisions 173593 via svnmerge from mmichelson1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r173593 | mmichelson | 2009-02-05 12:48:55 -0600 (Thu, 05 Feb 2009) | 11 lines Merged revisions 173592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb 2009) | 3 lines Add some missing cleanup to app_mixmonitor ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173594 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05Merged revisions 173589 via svnmerge from mmichelson1-5/+88
https://origsvn.digium.com/svn/asterisk/trunk ................ r173589 | mmichelson | 2009-02-05 12:34:06 -0600 (Thu, 05 Feb 2009) | 33 lines Merged revisions 173559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up. app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed audio to a file. Since this thread runs independently of the channel, it is possible that the mixmonitor thread's channel pointer will point to freed memory when the channel either is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the cases slightly differently). The solution for this is to employ a datastore, which has the nice benefit of allowing us to hook into channel masquerades and hangups and update our pointer as necessary. If this looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more involved since it does a lot more operations on the channel that is being spied upon. app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em- ploy a condition-and-boolean combination to ensure that the channel thread finishes with our structure before the mixmonitor thread attempts to free it. No crashes! (closes issue #14374) Reported by: aragon Patches: 14374.patch uploaded by putnopvut (license 60) Tested by: aragon, putnopvut ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173590 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05fix WORKING_FORK detectionjpeeler1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173554 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05regenerate with bootstrap.shtilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173548 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04I messed up and accidentally reverted the trunk-merged prop before ↵jpeeler0-0/+0
committing 173546. Added it manually. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173547 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04Merged revisions 173500 via svnmerge from jpeeler1-38/+65
https://origsvn.digium.com/svn/asterisk/trunk ................ r173500 | jpeeler | 2009-02-04 15:17:53 -0600 (Wed, 04 Feb 2009) | 23 lines Merged revisions 173211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009) | 17 lines Parking attempts made to one end of a bridge no longer will hang up due to a parking failure. Parking attempts made using either one-touch, or doing either a blind or assisted transfer to the parking extension now keep up the bridge instead of hanging up the attempted parked party. Normal causes for the parking attempt to fail includes the specific specified extension (via PARKINGEXTEN) not being available or if all the parking spaces are currently in use. To avoid having to reverse a masquerade park_space_reserve was made to provide foresight if a parking attempt will succeed and if so reserve the parking space. (closes issue #13494) Reported by: mdu113 Reviewed by Russell: http://reviewboard.digium.com/r/133/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173546 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04Merged revisions 173507 via svnmerge from mmichelson1-3/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r173507 | mmichelson | 2009-02-04 16:16:19 -0600 (Wed, 04 Feb 2009) | 7 lines Fix some areas where the incorrect interface was passed to ast_device_state I swear it feels like I already did this once... (closes issue #14359) Reported by: francesco_r ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173534 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04Merged revisions 173458 via svnmerge from tilghman1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r173458 | tilghman | 2009-02-04 12:48:06 -0600 (Wed, 04 Feb 2009) | 9 lines When using a socket as a FILE *, the stdio functions will sometimes try to do an fseek() on the stream, which is an invalid operation for a socket. Turning off buffering explicitly lets the stdio functions know they cannot do this, thus avoiding a potential error. (closes issue #14400) Reported by: fnordian Patches: tcptls.patch uploaded by fnordian (license 110) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173460 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04Merged revisions 173397 via svnmerge from mmichelson1-3/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r173397 | mmichelson | 2009-02-04 11:45:14 -0600 (Wed, 04 Feb 2009) | 11 lines Merged revisions 173396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb 2009) | 3 lines Revert my previous change because it was stupid ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173398 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04Merged revisions 173393 via svnmerge from mmichelson1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r173393 | mmichelson | 2009-02-04 11:41:02 -0600 (Wed, 04 Feb 2009) | 11 lines Merged revisions 173392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb 2009) | 3 lines Add a missing unlock. Extremely unlikely to ever matter, but it's needed. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173394 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04Merged revisions 173354 via svnmerge from mmichelson1-1/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r173354 | mmichelson | 2009-02-04 09:30:12 -0600 (Wed, 04 Feb 2009) | 30 lines Fix a problem where file playback would cause fds to remain open forever The problem came from the fact that a frame read from a format interpreter was not freed. Adding a call to ast_frfree fixed this. The explanation for why this caused the problem is a bit complex, but here goes: There was a problem in all versions of Asterisk where the embedded frame of a filestream structure was referenced after the filestream was freed. This was fixed by adding reference counting to the filestream structure. The refcount would increase every time that a filestream's frame pointer was pointing to an actual frame of data. When the frame was freed, the refcount would decrease. Once the refcount reached 0, the filestream was freed, and as part of the operation, the open files were closed as well. Thus it becomes more clear why a missing ast_frfree would cause a reference leak and cause the files to not be closed. You may ask then if there was a frame leak before this patch. The answer to that is actually no! The filestream code was "smart" enough to know that since the frame we received came from a format interpreter, the frame had no malloced data and thus didn't need to be freed. Now, however, there is cleanup that needs to be done when we finish with the frame, so we do need to call ast_frfree on the frame to be sure that the refcount for the filestream is decremented appropriately. (closes issue #14384) Reported by: fiddur Patches: 14384.patch uploaded by putnopvut (license 60) Tested by: fiddur, putnopvut ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173355 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04Merged revisions 173311 via svnmerge from tilghman1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r173311 | tilghman | 2009-02-03 18:43:52 -0600 (Tue, 03 Feb 2009) | 10 lines Ensure that commas placed in the middle of extension character classes do not interfere with correct parsing of the extension. Also, if an unterminated character class DOES make its way into the pbx core (through some other method), ensure that it does not crash Asterisk. (closes issue #14362) Reported by: Nick_Lewis Patches: 20090129__bug14362.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173312 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-03Fixes issue with IAX2 transfer not handing of calls. dvossel1-8/+74
Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. (issue #13468) Review: http://reviewboard.digium.com/r/140/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173250 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-03Merged revisions 173104 via svnmerge from tilghman1-1/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r173104 | tilghman | 2009-02-02 18:24:52 -0600 (Mon, 02 Feb 2009) | 12 lines Merged revisions 173070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009) | 5 lines Add warning to standard config, that globals may be overridden by other dialplan configuration files. (closes issue #14388) Reported by: macli ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173111 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-02Merged revisions 173067 via svnmerge from twilson1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r173067 | twilson | 2009-02-02 17:57:25 -0600 (Mon, 02 Feb 2009) | 9 lines Merged revisions 173066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009) | 2 lines Fix a feature inheritance bug I added after code review ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173068 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-02Merged revisions 172894 via svnmerge from lmadsen1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r172894 | lmadsen | 2009-02-02 13:13:40 -0500 (Mon, 02 Feb 2009) | 7 lines Update the res_ldap.conf file with a better working example. (closes issue #13861) Reported by: scramatte Patches: __20080110-res_ldap.conf-2.patch uploaded by blitzrage (license 10) Tested by: jcovert ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172896 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-02Blocked revisions 172890 via svnmergemurf0-0/+0
........ r172890 | murf | 2009-02-02 10:37:15 -0700 (Mon, 02 Feb 2009) | 41 lines This change allows the disconnect feature (as in "one-touch" in features.c) to be used within the dial app, before a call is bridged. Many thanks to sobomax for submitting this patch. Quoting from bug 11582: "So the goal of the patch was to use the user configured feature code during the call setup phase. The original ast_feature_interpret() function is not well suited for this purpose as it uses much call bridge specific data and doesn't separate a detection of feature from a feature handler call. So a new function ast_feature_detect() has been extracted off the ast_feature_interpret() function but keeping the original logic intact except some insignificant changes to locking. "Having created the ast_feature_detect() function the possibility to use feature detection in almost any place of the asterisk code. So a call to this function has been added to wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler however and uses old call leg disconnect logic to make the changes as small and simple as possible to prevent unexpected problems. A disconnect feature currently is the only one supported during call setup as other features as call parking and call transfer don't make much sense during call setup. However if need in some of the features would arise it is much easier to implement as the infrastructure changes are already in place with this patch." I have cleaned up the patch somewhat, and verified that the existing functionality is not harmed, and that the new functionality works. Terry has committed his stuff, and there were no conflicts (see 14274). (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) patch-include__asterisk__features.h uploaded by sobomax (license 359) patch-res__res_features.c uploaded by sobomax (license 359) enable-features-during-call-setup.diff uploaded by sobomax (license 359) 11583.newdiff uploaded by murf (license 17) enable-features-during-call-setup-1.diff uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) Tested by: sobomax, murf ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172892 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-01Merged revisions 172741 via svnmerge from tilghman1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r172741 | tilghman | 2009-01-31 20:44:23 -0600 (Sat, 31 Jan 2009) | 4 lines Blank argument crashes Asterisk (closes issue #14377) Reported by: amorsen ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172742 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-31Merged revisions 172706 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r172706 | tilghman | 2009-01-31 10:40:59 -0600 (Sat, 31 Jan 2009) | 7 lines Don't increment the loop, now that incrementing is taken care of by the decoder function. (closes issue #14363) Reported by: andrew53 Patches: func_strings_filter.patch uploaded by andrew53 (license 519) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172707 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-31Merged revisions 172581 via svnmerge from twilson1-2/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r172581 | twilson | 2009-01-30 15:50:03 -0600 (Fri, 30 Jan 2009) | 2 lines Remove incorret line from sample config ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172637 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-30Merged revisions 172580 via svnmerge from twilson6-116/+278
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/1.6.0@172635 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-30Merged revisions 172598 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r172598 | mmichelson | 2009-01-30 16:22:04 -0600 (Fri, 30 Jan 2009) | 3 lines Fix redefinition of flag in channel.h ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172604 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-29Merged revisions 172441 via svnmerge from tilghman12-11/+169
https://origsvn.digium.com/svn/asterisk/trunk ................ r172441 | tilghman | 2009-01-29 17:15:40 -0600 (Thu, 29 Jan 2009) | 16 lines Merged revisions 172438 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if Asterisk runs as a non-root user and the administrator does a 'restart now', Asterisk loses the ability to set QOS on packets. (closes issue #14004) Reported by: nemo Patches: 20090105__bug14004.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172503 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-29Merged revisions 172400 via svnmerge from rmudgett2-50/+423
https://origsvn.digium.com/svn/asterisk/trunk ........ r172400 | rmudgett | 2009-01-29 14:38:34 -0600 (Thu, 29 Jan 2009) | 12 lines channels/chan_dahdi.c * Added doxygen comments to the major dahdi structures. * Fixed PRI and SS7 using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed SS7 not checking if the dialed extension is at least as long as the stripmsd option. * Fixed PRI not handling unknown TON/NPI prefix letters correctly. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172434 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-29Merged revisions 172315 via svnmerge from tilghman1-12/+23
https://origsvn.digium.com/svn/asterisk/trunk ........ r172315 | tilghman | 2009-01-29 10:48:25 -0600 (Thu, 29 Jan 2009) | 2 lines Better document mode=multirow, based upon a conversation with Jared. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172316 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-29Merged revisions 172271 via svnmerge from lmadsen1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r172271 | lmadsen | 2009-01-29 08:47:27 -0500 (Thu, 29 Jan 2009) | 5 lines The realtime_pgsql.sql script is missing a couple of fields. closes issue #14339) Reported by: fiddur Patches: realtime_pgsql.sql.diff uploaded by fiddur (license 678) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172273 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-29Merged revisions 172173 via svnmerge from oej1-6/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r172173 | oej | 2009-01-29 10:18:01 +0100 (Tor, 29 Jan 2009) | 24 lines Merged revisions 172169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 lines Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause. This patch implements a temporary storage in the pvt and use that instead. The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header) Thanks to Klaus Darillion for testing! (closes issue #14294) related to issue #13385 Reported by: klaus3000 and adomjan Patches: bug14294b.diff uploaded by oej (license 306) Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487) Tested by: oej, klaus3000 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172217 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-28Merged revisions 172063 via svnmerge from murf5-14/+59
https://origsvn.digium.com/svn/asterisk/trunk ................ r172063 | murf | 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) | 52 lines Merged revisions 172030 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines This patch fixes h-exten running misbehavior in manager-redirected situations. What it does: 1. A new Flag value is defined in include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the bridge hangup exten code not to run the h-exten there (nor publish the bridge cdr there). It will done at the pbx-loop level instead. 2. In the manager Redirect code, I set this flag on the channel if the channel has a non-null pbx pointer. I did the same for the second (chan2) channel, which gets run if name2 is set... and the first succeeds. 3. I restored the ending of the cdr for the pbx loop h-exten running code. Don't know why it was removed in the first place. 4. The first attempt at the fix for this bug was to place code directly in the async_goto routine, which was called from a large number of places, and could affect a large number of cases, so I tested that fix against a fair number of transfer scenarios, both with and without the patch. In the process, I saw that putting the fix in async_goto seemed not to affect any of the blind or attended scenarios, but still, I was was highly concerned that some other scenarios I had not tested might be negatively impacted, so I refined the patch to its current scope, and jmls tested both. In the process, tho, I saw that blind xfers in one situation, when the one-touch blind-xfer feature is used by the peer, we got strange h-exten behavior. So, I inserted code to swap CDRs and to set the HANGUP_DONT field, to get uniform behavior. 5. I added code to the bridge to obey the HANGUP_DONT flag, skipping both publishing the bridge CDR, and running the h-exten; they will be done at the pbx-loop (higher) level instead. 6. I removed all the debug logs from the patch before committing. 7. I moved the AUTOLOOP set/reset in the h-exten code in res_features so it's only done if the h-exten is going to be run. A very minor performance improvement, but technically correct. (closes issue #14241) Reported by: jmls Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17) Tested by: murf, jmls ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172065 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-28Merged revisions 171964 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r171964 | tilghman | 2009-01-28 11:27:40 -0600 (Wed, 28 Jan 2009) | 9 lines Merged revisions 171963 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28 Jan 2009) | 2 lines Clarify log message (suggested by manxpower on #asterisk-dev) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@171965 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-28Merged revisions 171838 via svnmerge from oej1-0/+19
https://origsvn.digium.com/svn/asterisk/trunk ................ r171838 | oej | 2009-01-28 14:11:44 +0100 (Ons, 28 Jan 2009) | 10 lines Merged revisions 171837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 lines Add a better explanation of the difference between the device namespace and the dialplan for newbies. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@171846 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-27Blocked revisions 171757 via svnmergedvossel0-0/+0
........ r171757 | dvossel | 2009-01-27 16:43:36 -0600 (Tue, 27 Jan 2009) | 7 lines Adding AES_ENCRYPT and AES_DECRYPT dialplan functions. (closes issue #14301) Reported by: amorsen review: http://reviewboard.digium.com/r/128/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@171758 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-27Merged revisions 171691 via svnmerge from mmichelson1-65/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r171691 | mmichelson | 2009-01-27 15:58:39 -0600 (Tue, 27 Jan 2009) | 47 lines Merged revisions 171689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan 2009) | 39 lines Fix devicestate problems for "always-on" agent channels A revision to chan_agent attempted to "inherit" the device state of the underlying channel in order to report the device state of an agent channel more accurately. The problem with the logic here is that it makes no sense to use this for always-on agents. If the agent is logged in, then to the underlying channel, the agent will always appear to be "in use," no matter if the agent is on a call or not. The reason is that to the underlying channel, the channel is currently in use on a call to the AgentLogin application. The most common cause that I found for this issue to occur was for a SIP channel to be the underlying channel type for an Agent channel. If the SIP phone re-registers, then the registration will cause the device state core to query the device state of the SIP channel. Since the SIP channel is in use, the Agent channel would also inherit this status. Once the agent channel was set to "in use" there was no way that the device state could change on that channel unless the agent logged out. The solution for this problem is a bit different in 1.4 than it is in the other branches. In 1.4, there will be a one-line fix to make sure that only callback agents will inherit device state from their underlying channel type. For the other branches of Asterisk, since callback support has been removed, there is also no need for device state inheritance in chan_agent, so I will simply be removing it from the code. In addition, the 1.4 source is getting a new comment to help the next person who edits chan_agent.c. I'm adding a comment that a agent_pvt's loginchan field may be used to determine if the agent is a callback agent or not. (closes issue #14173) Reported by: nathan Patches: 14173.patch uploaded by putnopvut (license 60) Tested by: nathan, aramirez ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@171692 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-27Merged revisions 171622 via svnmerge from mmichelson1-0/+10
https://origsvn.digium.com/svn/asterisk/trunk ................ r171622 | mmichelson | 2009-01-27 14:11:30 -0600 (Tue, 27 Jan 2009) | 26 lines Merged revisions 171621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan 2009) | 18 lines Prevent a crash from occurring when a jitter buffer interpolated frame is removed from a slinfactory slinfactory used the "samples" field of an ast_frame in order to determine the amount of data contained within the frame. In certain cases, such as jitter buffer interpolated frames, the frame would have a non-zero value for "samples" but have NULL "data" This caused a problem when a memcpy call in ast_slinfactory_read would attempt to access invalid memory. The solution in use here is to never feed frames into the slinfactory if they have NULL "data" (closes issue #13116) Reported by: aragon Patches: 13116.diff uploaded by putnopvut (license 60) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@171623 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-27Merged revisions 171618 via svnmerge from mmichelson1-16/+44
https://origsvn.digium.com/svn/asterisk/trunk ........ r171618 | mmichelson | 2009-01-27 13:30:54 -0600 (Tue, 27 Jan 2009) | 24 lines Fix queue crashes that would occur after the calling channel was masqueraded. The data passed to the end_bridge_callback was assumed to be data which was still stack'd. The problem was that with some call features, attended transfers in particular, a new bridge thread is started once the feature completes, meaning that when the end_bridge_callback is called, the end_bridge_callback_data was invalid. To fix this problem, there are two measures taken 1. Instead of pointing to stacked data, we now used heap-allocated data for passing to the end_bridge_callback in app_queue 2. Since bridges can end multiple times on a single logical call, we wait until the final bridge is broken to actually set any queue variables. This is accomplished through reference-counting and the use of an end_bridge_callback_data_fixup function in app_queue.c (closes issue #14260) Reported by: ccesario Patches: 14260.patch uploaded by putnopvut (license 60) Tested by: ccesario ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@171619 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-27Revert some changes that shouldn't have made it inmattf2-79/+79
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@171595 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-27Make sure we do not go into alarm on PTMP links with non persistent D-channelsmattf3-80/+84
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@171594 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-27Merged revisions 171528 via svnmerge from oej1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r171528 | oej | 2009-01-27 16:00:19 +0100 (Tis, 27 Jan 2009) | 23 lines Solving the same issue, but a bit different in trunk... Merged revisions 171527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13 lines Use the same branch tag in CANCEL as in INVITE Originally putnopvut implemented some changes in revision 142079 that according to the bug report seemed to have worked then, but somehow fails now. I guess code, as humans, get old and forget stuff. Anyway, this bug caused CANCEL not to work with picky systems. Thanks Fredrik for pointing out where the bug in the SIP messaging was. (closes issue #14346) Reported by: oej Patches: bug14346.diff uploaded by oej (license 306) Tested by: oej ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@171529 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-26Merged revisions 171326 via svnmerge from oej1-5/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r171326 | oej | 2009-01-26 14:44:40 +0100 (MÃ¥n, 26 Jan 2009) | 17 lines Merged revisions 171264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9 lines Don't retransmit 401 on REGISTER requests when alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000 Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@171327 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-26Merged revisions 171188 via svnmerge from tilghman1-10/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r171188 | tilghman | 2009-01-25 17:58:00 -0600 (Sun, 25 Jan 2009) | 13 lines Merged revisions 171187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009) | 6 lines Correctly track the hookstate (closes issue #13686) Reported by: itiliti Patches: 20081013__bug13686.diff.txt uploaded by Corydon76 (license 14) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@171189 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-25Blocked revisions 171081 via svnmergemvanbaak0-0/+0
........ r171081 | mvanbaak | 2009-01-25 17:50:53 +0100 (Sun, 25 Jan 2009) | 2 lines dont segfault when a MWI event occurs on a line without a registered device ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@171082 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-25Blocked revisions 171043 via svnmergemvanbaak0-0/+0
........ r171043 | mvanbaak | 2009-01-25 15:35:17 +0100 (Sun, 25 Jan 2009) | 7 lines Make the sample skinny.conf work (closes issue #14325) Reported by: DEA Patches: skinny.conf.sample-trunk.txt uploaded by DEA (license 3) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@171044 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-25Merged revisions 170980 via svnmerge from seanbright1-4/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r170980 | seanbright | 2009-01-25 08:35:48 -0500 (Sun, 25 Jan 2009) | 16 lines Merged revisions 170979 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan 2009) | 9 lines Resolve a logic error that was causing Page() to crash when more than one channel was specified. (closes issue #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt uploaded by seanbright (license 71) Tested by: kc0bvu ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@170981 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-25Merged revisions 170943 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r170943 | russell | 2009-01-24 20:49:30 -0600 (Sat, 24 Jan 2009) | 6 lines Change ARRAY_LEN() to be more C++ safe. When the second part of this macro is written as 0[a] instead of a[0], it will force a failure if the macro is used on a C++ object that overloads the [] operator. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@170944 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-24Blocked revisions 170902 via svnmergerussell0-0/+0
........ r170902 | russell | 2009-01-24 13:33:15 -0600 (Sat, 24 Jan 2009) | 2 lines Add a todo to finish the XML docs in this module ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@170903 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-24Merged revisions 170837 via svnmerge from tilghman1-5/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r170837 | tilghman | 2009-01-24 07:55:53 -0600 (Sat, 24 Jan 2009) | 9 lines Merged revisions 170836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24 Jan 2009) | 2 lines Remove superfluous implementation note (closes issue #14319) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@170838 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-23Merged revisions 170794 via svnmerge from rmudgett1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r170794 | rmudgett | 2009-01-23 17:10:34 -0600 (Fri, 23 Jan 2009) | 1 line Fix asterisk.pdf generation if branch name has an underscore in it. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@170830 f38db490-d61c-443f-a65b-d21fe96a405b
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