285 Commits (910aaac5aed0a9a27c97ffd74fe7c02ad8e210a2)

Author	SHA1	Message	Date
Joshua Colp	642b571683	Merged revisions 203699 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines Improve T.38 negotiation by exchanging session parameters between application and channel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@203703 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	f46fa522ad	Merged revisions 202497 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r202497 | russell | 2009-06-22 15:11:04 -0500 (Mon, 22 Jun 2009) | 11 lines Merged revisions 202496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009) | 4 lines Report CallerID change during a masquerade. Reported by: markster ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@202503 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	311e463f48	Merged revisions 201458 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r201458 | mmichelson | 2009-06-17 15:04:12 -0500 (Wed, 17 Jun 2009) | 15 lines Merged revisions 201450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun 2009) | 9 lines Change the datastore traversal in ast_do_masquerade to use a safe list traversal. It is possible for datastore fixup functions to remove the datastore from the list and free it. In particular, the queue_transfer_fixup in app_queue does this. While I don't yet know of this causing any crashes, it certainly could. Found while discussing a separate issue with Brian Degenhardt. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201460 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	bbfd73e692	Merged revisions 201056 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun 2009) | 18 lines Merged revisions 200991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines Improve support for media paths that can generate multiple frames at once. There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	e0aa37c39a	Merged revisions 200361 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun 2009) | 16 lines Merged revisions 200360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun 2009) | 10 lines Suppress a warning message and give a better return code when generating inband ringing after a call is answered. (closes issue #15158) Reported by: madkins Patches: 15158.patch uploaded by mmichelson (license 60) Tested by: madkins ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	935853d4a3	Merged revisions 198856 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 | dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines Generic call forward api, ast_call_forward() The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options. (closes issue #13630) Reported by: festr Review: https://reviewboard.asterisk.org/r/271/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	c1df063c75	Merged revisions 198072 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May 2009) | 21 lines Merged revisions 198068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition. This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels. (closes issue #12946) Reported by: meral Patches: null-cdr2.diff uploaded by mnicholson (license 96) Tested by: mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested by: sum ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198074 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	538fc9986c	Fix broken attended transfers ... The bridge was terminating immediately after the attended transfer was completed. The problem was because upon reentering ast_channel_bridge nexteventts was checked to see if it was set and if so could possibly return AST_BRIDGE_COMPLETE. (closes issue #15183) Reported by: andrebarbosa Tested by: andrebarbosa, tootai, loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@197145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	7c4e45ab6e	Recorded merge of revisions 194357 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r194357 | mmichelson | 2009-05-13 14:42:51 -0500 (Wed, 13 May 2009) | 18 lines Blocked revisions 194356 via svnmerge ........ r194356 | mmichelson | 2009-05-13 14:41:44 -0500 (Wed, 13 May 2009) | 13 lines Remove an extraneous unlocking operation from ast_channel_free. In the case that we could not remove the desired channel from the list of channels, there was an extra call to unlock the channel list. Since we unlock the list later on in the function anyway, this results in the list being unlocked twice yet only being locked once. (closes issue #15098) Reported by: tim_ringenbach Patches: remove_extra_unlock.diff uploaded by tim (license 540) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@194359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	1e4552ef29	Merged revisions 192318 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r192318 | kpfleming | 2009-05-05 12:34:19 +0200 (Tue, 05 May 2009) | 5 lines Properly account for memory allocated for channels and datastores As in previous commits, when channels are allocated (with ast_channel_alloc) or datastores are allocated (with ast_datastore_alloc) properly account for the memory being owned by the caller, instead of the allocator function itself. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@192354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	e9bca89fc7	Merged revisions 191489 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r191489 | jpeeler | 2009-05-01 13:09:23 -0500 (Fri, 01 May 2009) | 15 lines Merged revisions 191488 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009) | 9 lines Fix DTMF not being sent to other side after a partial feature match This fixes a regression from commit 176701. The issue was that ast_generic_bridge never exited after the feature digit timeout had elapsed, which prevented the queued DTMF from being sent to the other side. This issue was reported to me directly. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@191505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	128d8e31b7	Merged revisions 189278 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r189278 | mmichelson | 2009-04-20 09:05:27 -0500 (Mon, 20 Apr 2009) | 18 lines Merged revisions 189277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr 2009) | 12 lines Move the check for chan->fdno == -1 to after the zombie/hangup check. Many users were finding that their hung up channels were staying up and causing 100% CPU usage. (issue #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch uploaded by mmichelson (license 60) Tested by: falves11, bamby ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	d470a900fa	Merged revisions 187599 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r187599 | tilghman | 2009-04-09 22:55:27 -0500 (Thu, 09 Apr 2009) | 2 lines Modify headers and macros, according to Russell's suggestions on the -dev list ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@187600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	278f2646a6	Merged revisions 186985 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r186985 | mmichelson | 2009-04-08 10:27:41 -0500 (Wed, 08 Apr 2009) | 30 lines Merged revisions 186984 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr 2009) | 24 lines Make a couple of changes with regards to a new message printed in ast_read(). "ast_read() called with no recorded file descriptor" is a new message added after a bug was discovered. Unfortunately, it seems there are a bunch of places that potentially make such calls to ast_read() and trigger this error message to be displayed. This commit does two things to help to make this message appear less. First, the message has been downgraded to a debug level message if dev mode is not enabled. The message means a lot more to developers than it does to end users, and so developers should take an effort to be sure to call ast_read only when a channel is ready to be read from. However, since this doesn't actually cause an error in operation and is not something a user can easily fix, we should not spam their console with these messages. Second, the message has been moved to after the check for any pending masquerades. ast_read() being called with no recorded file descriptor should not interfere with a masquerade taking place. This could be seen as a simple way of resolving issue #14723. However, I still want to try to clear out the existing ways of triggering this message, since I feel that would be a better resolution for the issue. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@186987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	5fb82960ab	Merged revisions 186833 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r186833 | mmichelson | 2009-04-07 18:50:56 -0500 (Tue, 07 Apr 2009) | 15 lines Merged revisions 186832 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr 2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a p2p bridge returns AST_BRIDGE_RETRY. Without this flag set, warning sounds will not be properly played to either party of the bridge. (closes issue #14845) Reported by: adomjan ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@186835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5d677e2d39	Merged revisions 185772 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r185772 | russell | 2009-04-01 08:48:26 -0500 (Wed, 01 Apr 2009) | 14 lines Merged revisions 185771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009) | 6 lines Fix a case where DTMF could bypass audiohooks. This change fixes a situation where an audiohook that wants DTMF would not actually get it. This is in the code path where we end DTMF digit length emulation while handling a NULL frame. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@185774 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	ad618c6c4f	Merged revisions 184762 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r184762 | kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar 2009) | 12 lines Improve timing interface to remember which provider provided a timer The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error. This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider. (closes issue #14697) Reported by: moy Review: http://reviewboard.digium.com/r/211/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@184765 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	baa195703b	Merged revisions 183057 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r183057 | file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines Fix an issue where a T38 control frame would get dropped. If two channels were bridged together using a generic bridge the T38 control frame would get passed up instead of being indicated on the other channel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@183067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	baab6e74b9	Merged revisions 182847 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@182946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	d2f5ee0407	Merged revisions 182553 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r182553 | russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines Tweak the handling of the frame list inside of ast_answer(). This does not change any behavior, but moves the frames from the local frame list back to the channel read queue using an O(n) algorithm instead of O(n^2). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@182570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	860b5570e3	Merged revisions 182530 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r182530 | kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2 lines correct logic flaw in ast_answer() changes in r182525 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@182533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	4d1d39c1e7	Merged revisions 182525 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 | kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11 lines Improve behavior of ast_answer() to not lose incoming frames ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations. When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames. This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller. http://reviewboard.digium.com/r/196/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@182527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	d52bcf4383	Merged revisions 182171 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r182171 | file | 2009-03-16 10:58:24 -0300 (Mon, 16 Mar 2009) | 7 lines Fix a memory leak in the ast_answer / __ast_answer API call. For a channel that is not yet answered this API call will wait until a voice frame is received on the channel before returning. It does this by waiting for frames on the channel and reading them in. The frames read in were not freed when they should have been. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@182173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	e9ed2c1920	Merged revisions 181465 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r181465 | russell | 2009-03-11 17:25:57 -0500 (Wed, 11 Mar 2009) | 2 lines Make handling of the BRIDGE_PLAY_SOUND variable thread-safe. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@181474 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	846b799e5b	Merged revisions 181428 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r181428 | russell | 2009-03-11 17:14:55 -0500 (Wed, 11 Mar 2009) | 2 lines Make handling of the BRIDGEPVTCALLID variable thread-safe. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@181430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	d46ccc57ab	Merged revisions 181424 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r181424 | russell | 2009-03-11 16:49:29 -0500 (Wed, 11 Mar 2009) | 17 lines Merged revisions 181423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) | 9 lines Make code that updates BRIDGEPEER variable thread-safe. It is not safe to read the name field of an ast_channel without the channel locked. This patch fixes some places in channel.c where this was being done, and lead to crashes related to masquerades. (closes issue #14623) Reported by: guillecabeza ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@181426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	84b495160a	Merged revisions 180032 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r180032 | dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines app_read does not break from prompt loop with user terminated empty string In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata(). (closes issue #14279) Reported by: Marquis Patches: fix_app_read.patch uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@180080 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	14ac305777	Merged revisions 179742 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009) | 14 lines Merged revisions 179741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) | 6 lines Ensure chan->fdno always gets reset to -1 after handling a channel fd event. Since setting fdno to -1 had to be moved, a couple of other code paths that do process an fd event return early and do not pass through the code path where it was moved to. So, set it to -1 in a few other places, too. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@179744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	064f510266	Merged revisions 179672 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) | 10 lines Merged revisions 179671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3 lines Move where fdno is set to the default value to *after* the read callback of the channel driver is called. We have to do this as the underlying channel driver may need the fdno value to determine what to read. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@179674 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	e20075c14a	Merged revisions 179609 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009) | 17 lines Merged revisions 179608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) | 9 lines Make it easier to detect an improper call to ast_read(). When you call ast_waitfor() on a channel, the index into the channel fds array that holds the file descriptor that poll() determines has input available is stored in fdno. This patch clears out this value after a call to ast_read() and also reports errors if ast_read() is called without an fdno set. From a discussion on the asterisk-dev list. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@179611 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	97c4d8f7aa	Merged revisions 179537 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009) | 21 lines Merged revisions 179536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) | 15 lines Fix bridging regression from commit 176701 This fixes a bad regression where the bridge would exit after an attended transfer was made. The problem was due to nexteventts getting set after the masquerade which caused the bridge to return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: tim_ringenbach ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@179539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	79d21b7843	Merged revisions 179462 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009) | 16 lines Merged revisions 179461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) | 8 lines Ensure that only one thread is calling ast_settimeout() on a channel at a time. For example, with an IAX2 channel, you can have both the channel thread and the chan_iax2 processing threads calling this function, and doing so twice at the same time is a bad thing. (Found in a debugging session with dvossel and mmichelson) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@179464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	22dafa5396	Merged revisions 176708 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r176708 | jpeeler | 2009-02-17 16:08:00 -0600 (Tue, 17 Feb 2009) | 23 lines Merged revisions 176701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) | 17 lines Modify bridging to properly evaluate DTMF after first warning is played The main problem is currently if the Dial flag L is used with a warning sound, DTMF is not evaluated after the first warning sound. To fix this, a flag has been added in ast_generic_bridge for playing the warning which ensures that if a scheduled warning is missed, multiple warrnings are not played back (due to a feature evaluation or waiting for digits). ast_channel_bridge was modified to store the nexteventts in the ast_bridge_config structure as that information was lost every time ast_channel_bridge was reentered, causing a hangup due to incorrect time calculations. (closes issue #14315) Reported by: tim_ringenbach Reviewed on reviewboard: http://reviewboard.digium.com/r/163/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@176711 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	18b14510f2	Merged revisions 176666 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r176666 | russell | 2009-02-17 15:22:40 -0600 (Tue, 17 Feb 2009) | 16 lines Update the timing API to have better support for multiple timing interfaces. 1) Add module use count handling so that timing modules can be unloaded. 2) Implement unload_module() functions for the timing interface modules. 3) Allow multiple timing modules to be loaded, and use the one with the highest priority value. 4) Report which timing module is being use in the "timing test" CLI command. (closes issue #14489) Reported by: russell Review: http://reviewboard.digium.com/r/162/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@176675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	20655a3a05	Merged revisions 174945 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@174947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	4a89cdc0d5	Merged revisions 174844 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r174844 | file | 2009-02-11 10:44:47 -0400 (Wed, 11 Feb 2009) | 10 lines Tell the device state core a change happened when a channel is freed but not a specific state. We need to do this because while we know that the freeing of the channel may cause something to become not in use we do not know this for sure. There may be another channel that is still up which would cause it to be in use. (closes issue #13238) Reported by: kowalma Patches: 20090121__bug13238.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@174846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	c3711994a1	Merged revisions 170652 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r170652 | file | 2009-01-23 16:18:05 -0400 (Fri, 23 Jan 2009) | 11 lines Merged revisions 170648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4 lines When a channel is answered make sure any indications currently playing stop. Usually the phone would do this but if the channel was already answered then they are being generated by Asterisk and we darn well need to stop them. (closes issue #14249) Reported by: RadicAlish ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@170664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	94c4df06b0	Merged revisions 170393 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r170393 | mmichelson | 2009-01-23 09:44:27 -0600 (Fri, 23 Jan 2009) | 36 lines Merged revisions 170392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan 2009) | 28 lines Fix broken call pickup There was a subtle change in ast_do_masquerade which resulted in failed attempts to pickup calls. The problem was that the value of the AST_FLAG_OUTGOING flag was copied from the clone to the original channel. In the case of call pickup, this meant that the AST_FLAG_OUTGOING flag ended up being cleared on the channel that was attempting to execute the pickup. Because this flag was not set, when ast_read came across an answer frame, it ignored it. The result of this was that the calling channel was never properly answered. This fix changes the behavior in ast_do_masquerade to set the flags on the original channel to the union of the flags on the clone channel. This way, if the AST_FLAG_OUTGOING flag is set on either of the two channels involved in the masquerade, the resulting channel will have the flag set as well. (closes issue #14206) Reported by: francesco_r Patches: 14206.patch uploaded by putnopvut (license 60) Tested by: francesco_r, aragon, putnopvut ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@170395 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	a29672060a	Merged revisions 168562 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r168562 | russell | 2009-01-13 13:22:13 -0600 (Tue, 13 Jan 2009) | 10 lines Merged revisions 168561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines Revert unnecessary indications API change from rev 122314 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@168565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	63f2d52453	Merged revisions 166569 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r166569 | mmichelson | 2008-12-23 09:17:54 -0600 (Tue, 23 Dec 2008) | 20 lines Merged revisions 166568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec 2008) | 12 lines Fix a crash resulting from a datastore with inheritance but no duplicate callback The fix for this is to simply set the newly created datastore's data pointer to NULL if it is inherited but has no duplicate callback. (closes issue #14113) Reported by: francesco_r Patches: 14113.patch uploaded by putnopvut (license 60) Tested by: francesco_r ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166571 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	9981739dd1	Merged revisions 166533 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r166533 | tilghman | 2008-12-22 22:32:15 -0600 (Mon, 22 Dec 2008) | 11 lines Merged revisions 166509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008) | 4 lines Use the integer form of condition for integer comparisons. (closes issue #14127) Reported by: andrew ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166535 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	7ed70a7a3f	Merged revisions 166092,166095 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r166092 | mmichelson | 2008-12-19 16:26:16 -0600 (Fri, 19 Dec 2008) | 28 lines Adding a new dialplan function AUDIOHOOK_INHERIT This function is being added as a method to allow for an audiohook to move to a new channel during a channel masquerade. The most obvious use for such a facility is for MixMonitor when a transfer is performed. Prior to the addition of this functionality, if a channel running MixMonitor was transferred by another party, then the recording would stop once the transfer had completed. By using AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the call even after the transfer has completed. It has also been determined that since this is seen by most as a bug fix and is not an invasive change, this functionality will also be backported to 1.4 and merged into the 1.6.0 branches, even though they are feature-frozen. (closes issue #13538) Reported by: mbit Patches: 13538.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/102/ ........ r166095 | mmichelson | 2008-12-19 16:40:57 -0600 (Fri, 19 Dec 2008) | 5 lines Remove the verbatim tag from the author line I could have sworn I already did that before, though... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	e74bc96b7b	Merged revisions 164203 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r164203 | russell | 2008-12-15 08:40:24 -0600 (Mon, 15 Dec 2008) | 39 lines Merged revisions 164201 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008) | 31 lines Handle a case where a call can be bridged to a channel that is still ringing. The issue that was reported was about a case where a RINGING channel got redirected to an extension to pick up a call from parking. Once the parked call got taken out of parking, it heard silence until the other side answered. Ideally, the caller that was parked would get a ringing indication. This patch fixes this case so that the caller receives ringback once it comes out of parking until the other side answers. The fixes are: - Make sure we remember that a channel was an outgoing channel when doing a masquerade. This prevents an erroneous ast_answer() call on the channel, which causes a bogus 200 OK to be sent in the case of SIP. - Add some additional comments to explain related parts of code. - Update the handling of the ast_channel visible_indication field. Storing values that are not stateful is pointless. Control frames that are events or commands should be ignored. - When a bridge first starts, check to see if the peer channel needs to be given ringing indication because the calling side is still ringing. - Rework ast_indicate_data() a bit for the sake of readability. (closes issue #13747) Reported by: davidw Tested by: russell Review: http://reviewboard.digium.com/r/90/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@164300 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	79d555f70c	Merged revisions 163579 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r163579 | file | 2008-12-12 12:55:15 -0400 (Fri, 12 Dec 2008) | 4 lines Since chan_sip is callback devicestate driven do not pass in actual states, pass in unknown so we get asked. Additionally do not pass in an actual device state value in ast_setstate since the channel may be callback driven. (closes issue #13525) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@163581 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	85ad905561	Merged revisions 163449 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r163449 | russell | 2008-12-12 07:55:30 -0600 (Fri, 12 Dec 2008) | 34 lines Merged revisions 163448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) | 26 lines Resolve issues that could cause DTMF to be processed out of order. These changes come from team/russell/issue_12658 1) Change autoservice to put digits on the head of the channel's frame readq instead of the tail. If there were frames on the readq that autoservice had not yet read, the previous code would have resulted in out of order processing. This required a new API call to queue a frame to the head of the queue instead of the tail. 2) Change up the processing of DTMF in ast_read(). Some of the problems were the result of having two sources of pending DTMF frames. There was the dtmfq and the more generic readq. Both were used for pending DTMF in various scenarios. Simplifying things to only use the frame readq avoids some of the problems. 3) Fix a bug where a DTMF END frame could get passed through when it shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation, and a digit arrived before emulation was complete, digits would get processed out of order. (closes issue #12658) Reported by: dimas Tested by: russell, file Review: http://reviewboard.digium.com/r/85/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@163515 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	39a8f051b7	Merged revisions 163171 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r163171 | russell | 2008-12-11 14:07:47 -0600 (Thu, 11 Dec 2008) | 16 lines Fix the "failed" extension for outgoing calls. The conversion to use ast_check_hangup() everywhere instead of checking the softhangup flag directly introduced this problem. The issue is that ast_check_hangup() checked for tech_pvt to be NULL. Unfortunately, this will be NULL is some valid circumstances, such as with a dummy channel. The fix is simple. Don't check tech_pvt. It's pointless, because the code path that sets this to NULL is when the channel hangup callback gets called. This happens inside of ast_hangup(), which is the same function responsible for freeing the channel. Any code calling ast_check_hangup() better not be calling it after that point, and if so, we have a bigger problem at hand. (closes issue #14035) Reported by: erogoza ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@163173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	9edc58f39e	Merged revisions 154687 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r154687 | murf | 2008-11-05 09:11:11 -0700 (Wed, 05 Nov 2008) | 9 lines Merged revisions 154685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154685 | murf | 2008-11-05 09:06:53 -0700 (Wed, 05 Nov 2008) | 1 line This fix was prompted by communication from user, who was seeing thousands of error logs... looks like EAGAIN. Made such uninteresting. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@154690 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 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	18 years ago
Russell Bryant	50c690fb59	Merged revisions 141949 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r141949 | russell | 2008-09-08 20:47:56 -0500 (Mon, 08 Sep 2008) | 9 lines Modify ast_answer() to not hold the channel lock while calling ast_safe_sleep() or when calling ast_waitfor(). These are inappropriate times to hold the channel lock. This is what has caused "could not get the channel lock" messages from chan_sip and has likely caused a negative impact on performance results of SIP in Asterisk 1.6. Thanks to file for pointing out this section of code. (closes issue #13287) (closes issue #13115) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@141951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	aaff8a2a20	Merged revisions 141157 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r141157 | murf | 2008-09-05 08:18:43 -0600 (Fri, 05 Sep 2008) | 9 lines Merged revisions 141156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1 line A small change to prevent double-posting of CDR's; thanks to Daniel Ferrer for bringing it to our attention ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@141159 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago