1943 Commits (48522988ab2ff7b73a327eac20b248c02e6ef981)

Author	SHA1	Message	Date
Sean Bright	48522988ab	In order to move away from nested function use, some changes to the recently introduced ... ast_channel_search_locked need to be made. Specifically, the caller needs to be able to pass arbitrary data which in turn is passed to the callback. This patch addresses all of the nested functions currently in asterisk trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155590 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	9ef09ad1d4	Merged revisions 155553 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov 2008) | 6 lines Use static functions here instead of nested ones. This requires a small change to the ast_bridge_config struct as well. To understand the reason for this change, see the following post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	30d1744ffc	Add ability to pass arbitrary data to the ao2_callback_fn (called from ... ao2_callback and ao2_find). Currently, passing OBJ_POINTER to either of these mandates that the passed 'arg' is a hashable object, making searching for an ao2 object based on outside criteria difficult. Reviewed by Russell and Mark M. via ReviewBoard: http://reviewboard.digium.com/r/36/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155401 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	1d09d193e7	Convert open-coded linked list in indications to the AST_LIST_* macros. This ... cleans the code up some and should make it more maintainable as time goes on. Reviewed by Russell, Kevin, Mark M., and Tilghman via ReviewBoard: http://reviewboard.digium.com/r/34/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	1a239454f1	Fix some code in chan_sip that was intended to unlink multiple objects from a ... container. The OBJ_MULTIPLE flag must be provided here. Otherwise, this would only remove a single object. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	9cc7bc998b	If 'asterisk.conf' is not found, instead of giving up, ... load documentation for the 'en_US' language (fix my last commit). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	65d4d1eb0f	Fix an asterisk crash if no asterisk.conf configuration file is present. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	e771f08f60	Simplify the output of [See Also]. ... Functions are printed without parenthesis like: FUNTION Applications are printed with parenthesis like: AppName() Cli commands are printed like: 'core show application' The other type of references are printed as they are inside the <ref> tag. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154967 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	6ac794074e	Update a couple places to use the new ast_channel_search_locked API call. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154923 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	81fb7597e5	Don't read history on -rx commands. ... (Closes issue #13571) Reported by: tzafrir Patch '0001-no-need-for-history-on-asterisk-rx.patch' uploaded by tzafrir. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154922 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	0d25ddd366	Add LISTFILTER dialplan function, along with supporting documentation. See ... documentation for more information on how to use it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154915 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	f7c20e0dec	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/trunk@154687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	990a6bebe8	Add more SeeAlso references based on TFOT. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	f18699be24	- Add more <see-also> based on TFOT. ... - Add the 'filename' type to the see-also ref. To be able to reference a filename. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	086a52d9d1	Introduce a new API call ast_channel_search_locked, which iterates through the ... channel list calling a caller-defined callback. The callback returns non-zero if a match is found. This should speed up some of the code that I committed earlier today in chan_sip (which is also updated by this commit). Reviewed by russellb and kpfleming via ReviewBoard: http://reviewboard.digium.com/r/28/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154429 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	2cc8e25222	Slightly optimize ast_devstate_str and rename global functions devstate2str and config_text_file_save to have an ast_ prefix ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	f349f18eaa	GLOB_BRACE is already added to MY_GLOB_FLAGS if it is supported on the ... platform. This should resolve some build errors on Solaris. (issue #13704) Reported by: dougm git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154191 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	bc1629a9e8	Fix build errors. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154186 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	fd6ee6e1f2	Merged revisions 154060 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008) | 3 lines Remove the potential for a division by zero error. (Closes issue #13810) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154061 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	bd4eb070f3	bring over all the fixes for the warnings found by gcc 4.3.x from the 1.4 branch, and add the ones needed for all the new code here too ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153616 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5b168ee34b	Merge changes from team/group/appdocsxml ... This commit introduces the first phase of an effort to manage documentation of the interfaces in Asterisk in an XML format. Currently, a new format is available for applications and dialplan functions. A good number of conversions to the new format are also included. For more information, see the following message to asterisk-dev: http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	d521ad9696	* Fixed timeout logic in the dialing API as setting timeouts ... had no effect * Updated dialing API documentation to indicate that timeouts are specified in milliseconds * Added a new timeout argument to the Page application. If time expires, any endpoints which have not answered will be hung up. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	5fe37e47c6	Recent CDR fixes moved execution of the 'h' exten into the bridging code, so variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call ... (closes issue #13793) Reported by: greenfieldtech git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153181 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	511ce6b2bf	Use the ast_str API call to reset the string instead of manually editing its internals ... (closes issue #13816) Reported by: eliel Patches: channel.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	85f78531ce	Merged revisions 152811 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152811 | kpfleming | 2008-10-30 11:53:48 -0500 (Thu, 30 Oct 2008) | 3 lines instead of comparing the string pointer to 0, let's compare the value that was actually parsed out of the string (found by sparse) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152812 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	10d36d9f34	fix a few small things found by using sparse ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	de90c84b1a	After seeing another problem in #asterisk stemming from ... the low default value of featuredigittimeout, I decided it was high time to change it. I have changed the default to 2000 ms based on a suggestion from Leif Madsen. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	fa06ce2e6c	Track down and fix annoying lock errors ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	6fad66dfb3	Merged revisions 152535 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines The magic trick to avoid this crash is not to try to find the channel by name in the list, which is slow and resource consuming, but rather to pay attention to the result codes from the ast_bridge_call, to which I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are returned when a channel is parked. Why? because CDR's aren't generated via parking, so nothing is needed, but if a transfer occurred, there are critical things I need. If you get AST_PBX_KEEPALIVE, then don't touch the channel pointer. If you get AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then don't touch the peer pointer. Updated the several places where the results from a bridge were not being properly obeyed, and fixed some code I had introduced so that the results of the bridge were not overridden (in trunk). All the places that previously tested for AST_PBX_NO_HANGUP_PEER now have to check for both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common parking scenarios: 1. A calls B; B answers; A parks B; B hangs up while A is getting the parking slot announcement, immediately after being put on hold. 2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but before the park times out. 3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold. 4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out. No crash. I also ran the scenarios above against valgrind, and accesses looked good. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	316f3897a8	Merged revisions 151905 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r151905 | russell | 2008-10-25 05:59:02 -0500 (Sat, 25 Oct 2008) | 8 lines Move AMI initialization to occur after loading modules. This prevents a deadlock when someone tries to initiate a module reload from the AMI just as Asterisk is starting. (closes issue #13778) Reported by: hotsblanc Fix suggested by hotsblanc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
BJ Weschke	9aefadd7c1	Do NOT attempt to do anything with the ast_config struct when it's been returned as INVALID by the config file interpreter. ... (closes issue #13741) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151246 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	1ddc834b39	cleaup of the TCP/TLS socket API: ... 1) rename 'struct server_args' to 'struct ast_tcptls_session_args', to follow coding guidelines 2) make ast_make_file_from_fd() static and rename it to something that indicates what it really is for (again coding guidelines) 3) rename address variables inside 'struct ast_tcptls_session_args' to be more descriptive (dare i say it... coding guidelines) 4) change ast_tcptls_client_start() to use the new 'remote_address' field of the session args for the destination of the connection, and use the 'local_address' field to bind() the socket to the proper source address, if one is supplied 5) in chan_sip, ensure that we pass in the PP address we are bound to when creating outbound (client) connections, so that our connections will appear from the correct address git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
BJ Weschke	09e9b5d208	Using the GetVar handler in AMI is potentially dangerous (insta-crash [tm]) when you use a dialplan function that requires a channel and then you don't provide one or provide an invalid one in the Channel: parameter. We'll handle this situation exactly the same way it was handled in pbx.c back on r61766. ... We'll create a bogus channel for the function call and destroy it when we're done. If we have trouble allocating the bogus channel then we're not going to try executing the function call at all and run the risk of crashing. (closes issue #13715) reported by: makoto patch by: bweschke git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150817 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	805556773f	Fix CLI command 'channel request hangup' ... Prodded on IRC by Russell and fixed by eliel (closes issue #13730) Reported by: eliel Patches: main_cli.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	4ad187cba4	Merged revisions 150304 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r150304 | mmichelson | 2008-10-16 18:40:54 -0500 (Thu, 16 Oct 2008) | 6 lines Reverting changes from commits 150298 and 150301 since I was mistakenly under the assumption that dialplan functions *always* required that a channel be present. I need to go home earlier, I think :) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150305 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	8a1d9d1678	Merged revisions 150298,150301 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r150298 | mmichelson | 2008-10-16 18:34:37 -0500 (Thu, 16 Oct 2008) | 10 lines Don't try to call a dialplan function's read callback from the manager's GetVar handler if an invalid channel has been specified. Several dialplan functions, including CHANNEL and SIP_HEADER, do not check for NULL-ness of the channel being passed in. (closes issue #13715) Reported by: makoto ........ r150301 | mmichelson | 2008-10-16 18:35:07 -0500 (Thu, 16 Oct 2008) | 3 lines And don't forget to return on the error condition ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	29a8fe20c8	Merged revisions 149204 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines Add a tolerance period for sync-triggered audiohooks so that if packetization of audio is close (but not equal) we don't end up flushing the audiohooks over small inconsistencies in synchronization. Related to issue #13005, and solves the issue for most people who were experiencing the problem. However, a small number of people are still experiencing the problem on long calls, so I am not closing the issue yet ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	d5837ba8c2	Add additional memory debugging to several core APIs, and fix several memory ... leaks found with these changes. (Closes issue #13505, closes issue #13543) Reported by: mav3rick, triccyx Patches: 20081001__bug13505.diff.txt uploaded by Corydon76 (license 14) Tested by: mav3rick, triccyx git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	b17413c992	Merged revisions 148611 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148611 | kpfleming | 2008-10-14 02:54:41 -0500 (Tue, 14 Oct 2008) | 3 lines it would be nice if this message printing code had actually been tested before it was committed... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	c1351ad237	Merge realtime_update2 branch, which adds a new realtime API call named ... 'update2', which permits updates which match across multiple columns, instead of requiring all tables to have a single unique identifier. All of the other API calls with the exception of 'update' already had the ability to match on multiple fields, so it was a missing and very desireable feature that an API call implementing an update should have this, too. This does not change any outward performance of Asterisk, but it should make life easier for application developers who use the RealTime framework. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	db7299f4bc	Hmmm. Nobody (but me) is interested in seeing ... the trie info when they do 'dialplan show ...' (even with debug set to non-zero); so I set up a 'dialplan debug [context]' cli command instead, to explicitly show just the trie info. I even added an extension_exists() call to make sure the trie info is built. I moved the explanatory header to above the extension loop to ensure it only prints once. And it will do this now, whether debug is set or not. I removed the trie printing from the 'dialplan show' command entirely. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	32d93bbc0e	Highlightning even more bugs in the current tcp/tls implementation. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	1dedb785ab	Don't include logger.h in asterisk.h by default as it is causing problems building ... app_voicemail. Instead, include it where it is needed. This turned out to be a relatively minor issue because other headers include logger.h as well. Need to test -addons before merging this back to 1.6.0. (closes issue #13605) Reported by: tomo1657 Patches: 13605_seanbright.diff uploaded by seanbright (license 71) Tested by: mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	9851feb8fb	The priority was unnecessary for the manager atxfer, so it has ... been removed. Furthermore, now we actually use the Context argument passed to set the transfer context and don't error out if no context is specified. This addresses the actual problems outlined in issue 12158. Regarding the other points brought up, regarding the inability to not transfer to extensions which cannot be represented by DTMF, it is not enough of a constraint that it is worth attempting to rework the feature. (closes issue #12158) Reported by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	6d70f45506	Merged revisions 146026 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) | 18 lines (closes issue #13579) Reported by: dwagner (closes issue #13584) Reported by: dwagner Tested by: murf, putnopvut The thought occurred to me that the res= from the extension spawn was ending up being returned from the bridge. "Thou shalt not poison the return value". Made the change and it appears to allow blind xfers to work as normal. If I'm wrong, reopen the bugs. But it looks good to me! Many thanks to putnopvut for helping me reproduce this! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	8b14e5f493	Reverting format addition for now ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f5d5eb5e19	Fudges for wav16, just like wav49 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	c897b4e630	(closes issue #13139) ... Reported by: krisk84 Tested by: krisk84 This change prevents a call that is placed in the parkinglot to be picked up before the PBX is finished. If another extension dials the parking extension before the PBX thread has completed at minimum warnings will occur about the PBX not properly being terminated. At worst, a crash could occur. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	e235a07376	(closes issue #13557) ... Reported by: nickpeirson Patches: pbx.c.patch uploaded by nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) Tested by: nickpeirson, murf 1. replaced all refs to bzero and bcopy to memset and memmove instead. 2. added a note to the CODING-GUIDELINES 3. add two macros to asterisk.h to prevent bzero, bcopy from creeping back into the source 4. removed bzero from configure, configure.ac, autoconfig.h.in git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	9335b3ad34	Allow people to select the old console behavior of white text on a black ... background, by using the startup flag '-B'. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago