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2007-11-29Fix build of trunkTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-29Merged revisions 90154 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90154 | tilghman | 2007-11-29 11:18:09 -0600 (Thu, 29 Nov 2007) | 2 lines Upgrade the core sounds release version ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90156 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-29let's try this again... *all* compilation and linking in Asterisk should be ↵Kevin P. Fleming
done using the standard compilation rules, not manually created ones. changing hashtest.c to use these rules caused the compiler to notice a large number of coding guidelines violations, so those are fixed too. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90150 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-29restore behavior from the 1.4 branch... manager users created via users.conf ↵Kevin P. Fleming
should default to *all* permissions, not none git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90149 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-29Blocked revisions 90147 via svnmergeRussell Bryant
........ r90147 | russell | 2007-11-28 18:36:59 -0600 (Wed, 28 Nov 2007) | 1 line fix some formatting i accidentally changed ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-29Merged revisions 90145 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90145 | russell | 2007-11-28 18:20:34 -0600 (Wed, 28 Nov 2007) | 5 lines This set of changes is to make some callerID handling thread-safe. The ast_set_callerid() function needed to lock the channel. Also, the handlers for the CALLERID() dialplan function needed to lock the channel when reading or writing callerid values directly on the channel structure. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-29Merged revisions 90142 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90142 | russell | 2007-11-28 18:06:08 -0600 (Wed, 28 Nov 2007) | 4 lines Merge a change from team/russell/chan_refcount ... This makes ast_stopstream() thread-safe. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-28Merge another small doxygen change from team/russell/chan_refcount to indicateRussell Bryant
that a channel doesn't need to be locked before calling a certain function. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-28Merge some channel.h doxygen updates from team/russell/chan_refcountRussell Bryant
This was mostly to note whether a channel needed to be locked or not before calling these functions. However, I added some other things, too. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-28Merged revisions 90101 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90101 | file | 2007-11-28 18:59:28 -0400 (Wed, 28 Nov 2007) | 6 lines Fix a few memory leaks. (closes issue #11405) Reported by: eliel Patches: load_realtime.patch uploaded by eliel (license 64) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-28Merged revisions 90098 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90098 | kpfleming | 2007-11-28 16:30:46 -0600 (Wed, 28 Nov 2007) | 2 lines it is impossible to set permissions for manager accounts created by users.conf (reported internally, patched by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90100 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-28file says... compile before you commit!Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90099 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-28Removing a pointless check of option_debugMark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-28Merged revisions 90059 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90059 | mmichelson | 2007-11-28 16:08:50 -0600 (Wed, 28 Nov 2007) | 13 lines Removing some seemingly pointless code. This sets a channel variable for every priority executed in the dialplan if you have debug set to anything non-zero. This seems pointless due to the fact that these channel variables are not referenced anywhere else in the code and their names are esoteric enough that they would not be practical to reference in the dialplan. Plus the fact that this behavior isn't documented anywhere means that the change is not likely to cause any disruption. If anything, this may actually cause a slight performance increase if running with debug on. The motivating influence for this code change is the eventwhencalled option for queues. If set to vars, all channel variables will be output to the manager. These unnecessary channel variables make the output a lot more difficult to deal with. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-28Made expr2 parser 8-bit transparentSteve Murphy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-28Remove "old"-style CLI handler, since nothing uses it anymore.Jason Parker
Closes issue #11403, patch by eliel. This also completes the janitor project. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-28Blocked revisions 89999 via svnmergeMark Michelson
........ r89999 | mmichelson | 2007-11-28 11:30:47 -0600 (Wed, 28 Nov 2007) | 6 lines Recording greetings when using IMAP storage was causing zero-length files to be stored. Since greetings are not retrieved from IMAP anyway, it is pointless to attempt storing them there. (closes issue #11359, reported by spditner, patched by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-28Hide CLI commands starting with _ from tab completion as was done previously.Joshua Colp
(closes issue #11395) Reported by: eliel Patches: cli.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-28Fix a few log messages.Joshua Colp
(closes issue #11396) Reported by: IgorG Patches: spell.v1.diff uploaded by IgorG (license 20) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89981 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-28Merge some little changes from team/russell/chan_refcount to help reduceRussell Bryant
the diff to trunk. This just removes some checks on the return value of alloca(), as behavior is undefined if it runs out of stack space, and we don't check it anywhere else. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-28Adding support for realtime music on hold. The following are the main points:Mark Michelson
1. When moh is started, we search first in memory to find the class. If we do not find it in memory, we search realtime instead. 2. When moh is restarted (as in, it had been started on this particular channel, stopped, and now we're starting it again), if using the "files" mode, then realtime will always be rechecked. If you are using other modes, however, we will simply reattach to the external running process which was playing moh earlier in the call. This is a necessary compromise so that we don't end up with too many background processes. 3. musiconhold.conf has a general section now. It has one option: cachertclasses. If set to yes, then moh classes found in realtime will be added to the in-memory list. This has the advantage of not requiring database lookups each time moh is started, but it has the disadvantage of not truly being realtime. I have tested this for functionality, and it passes. I also tested this under valgrind and there are no memory problems reported under typical use. Special thanks to Sergee for implementing this feature and enduring my complaints on the bugtracker! (closes issue #11196, reported and patched by sergee) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-28Merged revisions 89893 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89893 | russell | 2007-11-27 18:20:13 -0600 (Tue, 27 Nov 2007) | 4 lines - update documentation for some of the goto functions to note that they handle locking the channel as needed - update ast_explicit_goto() to lock the channel as needed ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Document that the channel is not locked when the send_digit_begin and endRussell Bryant
callbacks get called. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Merged revisions 89886 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89886 | russell | 2007-11-27 17:47:28 -0600 (Tue, 27 Nov 2007) | 2 lines Don't do frame processing if ast_read() returned NULL. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Merge changes from team/russell/iax2_frame_queueRussell Bryant
This patch is an optimization for chan_iax2. This module is now heavily multi-threaded. However, there is still a good number of globally shared resources that prevent things from happen asynchronously. One of those things was the global IAX frame queue. This queue was used to hold frames that have been deferred for transmitting by another thread, and frames that may need to get retransmitted. I changed the frame queue to be per-call, since almost all of the frame queue handling only cares about frames specific to a call number. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Merged revisions 89844 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89844 | russell | 2007-11-27 17:21:13 -0600 (Tue, 27 Nov 2007) | 3 lines Instead of depending on the return value of ast_true(), explicitly set the eventwhencalled variable to 1. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Merged revisions 89839 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89839 | russell | 2007-11-27 17:16:00 -0600 (Tue, 27 Nov 2007) | 2 lines Don't start/stop autoservice in pbx_extension_helper() unless a channel exists ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Merged revisions 89837 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89837 | mmichelson | 2007-11-27 17:10:05 -0600 (Tue, 27 Nov 2007) | 12 lines Two changes with regards to the 'eventwhencalled' option of queues.conf 1) Due to some signed vs. unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes' did exactly the same thing. Thus the sign change of the ast_true call. 2) The vars2manager function overwrote a \n for every channel variable it parsed, resulting in bizarre output for the channel variables. This patch remedies this. (related to issue #11385, however I'm not sure if this will actually be enough to close it) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Bring in a small change from team/russell/chan_refcountRussell Bryant
This replaces tab completion code with the use of a public function that does the same thing git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27closes issue #11294; missed the conditional unlock of the contexts when the ↵Steve Murphy
hash table is used instead; also, used the ast_free_ptr as advised. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Merged revisions 89790 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89790 | russell | 2007-11-27 15:45:51 -0600 (Tue, 27 Nov 2007) | 41 lines Merge changes from team/russell/autoservice_1.4 This set of changes fixes an issue that was reported to me on IRC yesterday. The user, d1mas, was using chan_zap for incoming calls and was having DTMF recognition issues in some situations. Specifically, he noticed that the problem occurred when using DISA or WaitExten. He also noticed that when using Read, the problem did not occur. His system also used DUNDi for dialplan lookups. So, he theorized that if the DUNDi lookups blocked for some period of time, that audio from the zap channel could get lost. If the audio got lost, then it wouldn't be run through the DTMF detector, and digits could get lost. He was correct, and the following set of changes fixes the problem. However, the changes go a little bit further than what was necessary to fix this exact problem. 1) I updated pbx_extension_helper() to autoservice the associated channel to handle cases where extension lookups may take a long time. This would normally be a dialplan switch that does some lookup over the network, such as the DUNDi or IAX2 switches. This ensures that even while a DUNDi lookup is blocking, the channel will be continuously serviced. 2) I made a change to the autoservice code. This is actually something that has bothered me for a long time. When a channel is in autoservice, _all_ frames get thrown away. However, some frames really shouldn't be thrown away. The most notable examples are signalling (CONTROL) frames, and DTMF. So, this patch queues up important frames while a channel is in autoservice. When autoservice is stopped on the channel, the queued up frames get stuck back on the channel so that they can get processed instead of thrown away. 3) I made another change to the autoservice code to handle the case where autoservice is started on channels recursively. Previously, you could call ast_autoservice_start() multiple times on a channel, and it would stop the first time ast_autoservice_stop() gets called. Now, it will ensure that autoservice doesn't actually stop until the final call to ast_autoservice_stop(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27A few more "moremanager" fixesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27More "moremanager" fixes. Manager commands to check module status.Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27More "moremanager" changes - doxygen docs and changing manager version (finally)Olle Johansson
before making more dramatic changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27More additions from the "moremanager" branch, this time for IAX2.Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Blocked revisions 89727 via svnmergeMark Michelson
........ r89727 | mmichelson | 2007-11-27 14:22:59 -0600 (Tue, 27 Nov 2007) | 6 lines Changing some calls from free() to ast_free() since they were allocated with ast_calloc(). (closes issue #11390, reported and patched by Laureano) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Merged revisions 89709 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89709 | kpfleming | 2007-11-27 14:16:56 -0600 (Tue, 27 Nov 2007) | 2 lines on second thought... revert all the other changes i've made in app options parsing leaving only one: if an empty argument is supplied for an option, set that argument pointer to point to an empty string rather than NULL, so that the application can do normal checks on it without worrying about it being NULL ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27remove a duplicate manager eventRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89710 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Manager events from the "moremanager" branchOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Merged revisions 89701 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89701 | kpfleming | 2007-11-27 13:36:55 -0600 (Tue, 27 Nov 2007) | 2 lines generate a warning when an application option that requires an argument is ignored due to lack of an argument ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89704 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Starting to merge changes from the "moremanager" branch. Documentation willOlle Johansson
follow. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89702 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27The following patch with updates for trunk. Works much better in trunk.Olle Johansson
Also by accident fixed a bad typo by a previous committer, which actually made video calls not work fully... Merged revisions 89630 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines If we get a codec offer using a well-known payload type, but using it for another codec that we don't know, Asterisk did not remove that codec from the list. With this patch, we remove the codec from audio and video rtp objects and deny it ever existed. Thanks to lasse for testing. (closes issue #11376) Reported by: lasse Patches: bug11376.txt uploaded by oej (license 306) Tested by: lasse ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Add an S_COR macro, which is similar to the existing S_OR macro,Jason Parker
except with an additional boolean arg. A hack such as: foo ? S_OR(bar, "baz") : "baz" becomes: S_COR(foo, bar, "baz") git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27made AEL 8-bit transparent; mainly the lexer was tossing chars with the ↵Steve Murphy
hi-order bit set. Not nice. Also, allow @ in extension names, and a backslash, also. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Ensure the value returned from ast_random is between 0 and RAND_MAX on ↵Joshua Colp
64-bit platforms. (closes issue #11348) Reported by: sperreault git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Merged revisions 89634 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89634 | russell | 2007-11-27 10:12:33 -0600 (Tue, 27 Nov 2007) | 3 lines Add a note to the sample voicemail config noting that when using IMAP storage, only the first format specified will be attached to the message. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Merged revisions 89631 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89631 | tilghman | 2007-11-27 09:38:03 -0600 (Tue, 27 Nov 2007) | 3 lines Default result of STAT should be "0" not "". Reported via the -users mailing list, fixed by me. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89632 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Merged revisions 89624 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89624 | oej | 2007-11-27 08:34:19 +0100 (Tis, 27 Nov 2007) | 6 lines Clarify limitonpeers=yes (closes issue #11304) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Merged revisions 89622 via svnmerge from Steve Murphy
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 line closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-26Change all instances of "CALLERID(number)" to "CALLERID(num)" forMark Michelson
consistency's sake (closes issue #11381, reported and patched by jon) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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