asterisk

Commit Graph

Author	SHA1	Message	Date
Tilghman Lesher	374e4e9d72	Fix bamboo compile error by calculating an integer with the same size as a pointer. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252980 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	385a40226a	Fix test_time on Mac OS X (and other platforms without inotify) Reviewboard: https://reviewboard.asterisk.org/r/554/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	68d1ded8dd	Only change the RTP ssrc when we see that it has changed This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	da6ba8e60e	Remove portions that weren't meant to be committed for the OS X compat fix git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	e58fc610ae	Change needed to make Mac OS X 10.6 happy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	925aad8944	Remove pbx_gtkconsole and related gtk1 checks. Review: https://reviewboard.asterisk.org/r/541/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	335558c5d1	Fix up the ast_rtp_property enum. The mis-placement of the latest entry meant that when it was set, it was writing one index past the end of the properties array in the ast_rtp_instance (which happened to be the local_address field). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250871 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	73ef4b8daf	Removed cdrflags from ast_channel structure. Only chan_dahdi set a value in cdrflags. Everyone else just copied it around the system. Noone cared about any value it may have contained. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	49aaeb1df2	Merge missed files from res_fax/res_fax_spandsp merge. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	06dc8bc123	Merge res_fax and res_fax_spandsp. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	862ebf4d00	fixes adaptive jitterbuffer configuration When configuring the adaptive jitterbuffer, the target_extra value not only could not be set from the configuration, but was not even being set to its proper default. This value is required in order for the adaptive jitterbuffer to work correctly. To resolve this a config option has been added to expose this value to the conf files, and a default value is provided when no config specific value is present. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	a6fd85250d	Properly document voicemail API documents. Also fix a crash reported via the -dev list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249405 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	53bf191569	Trim trailing whitespace, convert lists of defines to enums git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	4e1e6820db	Merged revisions 248582 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248582 \| tilghman \| 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010) \| 7 lines Remove color code sequences from verbose messages that go to logfiles. (closes issue #16786) Reported by: dodo Patches: logger2.patch uploaded by dodo (license 989) Tested by: tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	5665aa361f	Minor tweaks to comment blocks and includes. Fix the copyright lines, tweak doxygen formatting, and remove some unnecessary includes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	2ce7eabb24	Fix two problems in ast_str functions found while writing a unit test. 1. The documentation for ast_str_set and ast_str_append state that the max_len parameter may be -1 in order to limit the size of the ast_str to its current allocated size. The problem was that the max_len parameter in all cases was a size_t, which is unsigned. Thus a -1 was interpreted as UINT_MAX instead of -1. Changing the max_len parameter to be ssize_t fixed this issue. 2. Once issue 1 was fixed, there was an off-by-one error in the case where we attempted to write a string larger than the current allotted size to a string when -1 was passed as the max_len parameter. When trying to write more than the allotted size, the ast_str's __AST_STR_USED was set to 1 higher than it should have been. Thanks to Tilghman for quickly spotting the offending line of code. Oh, and the unit test that I referenced in the top line of this commit will be added to reviewboard shortly. Sit tight... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247335 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	164549691e	Add some clarifying documentation to the ast_str_set and ast_str_append functions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	1321748c19	Add a test module for the event API, test_event.c. This module includes a single test so far that creates events using two different methods and does some verification on the result to make sure the correct data can be retrieved from the event that was created. One bug was found in the event API while developing this test, which makes me happy. :-) Review: https://reviewboard.asterisk.org/r/495/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c8abb42e6a	Solaris doesn't like outputting a NULL to a %s in format strings. Detect all platforms that don't like that, either, and ensure that when documentation is missing, we pass a non-NULL pointer when outputting the corresponding documentation. (closes issue #16689) Reported by: bklang Patches: 20100209__issue16689__with_tests.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/497/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246030 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	bbed34f764	Various updates to the unit test API. 1) It occurred to me that the difference in usage between the error ast_str and the ast_test_update_status() usage has turned out to be a bit ambiguous in practice. In a lot of cases, the same message was being sent to both. In other cases, it was only sent to one or the other. My opinion now is that in every case, I think it makes sense to do both; we should output it to the CLI as well as save it off for logging purposes. This change results in most of the changes in this diff, since it required changes to all existing unit tests. It also allowed for some simplifications of unit test API implementation code. 2) Update ast_test_status_update() to include the file, function, and line number for the code providing the update. 3) There are some formatting tweaks here and there. Hopefully they aren't too distracting for code review purposes. Reviewboard's diff viewer seems to do a pretty good job of pointing out when something is a whitespace change. 4) I moved the md5_test and sha1_test into the test_utils module. It seemed like a better approach since these tests are so tiny. 5) I changed the number of nodes used in heap_test_2 from 1 million to 100 thousand. The only reason for this was to reduce the time it took for this test to run. 6) Remove an unused function prototype that was at the bottom of utils.h. 7) Simplify test_insert() using the LIST_INSERT_SORTALPHA() macro. The one minor difference in behavior is that it no longer checks for a test registered with the same name. 8) Expand the code in test_alloc() to provide specific error messages for each failure case, to clearly inform developers if they forget to set the name, summary, description, etc. 9) Tweak the output of the "test show registered" CLI command. I swapped the name and category to have the category first. It seemed more natural since that is the sort key. 10) Don't output the status ast_str in the "test show results" CLI command. This is going to tend to be pretty verbose, so just leave that for the detailed test logs (test generate results). Review: https://reviewboard.asterisk.org/r/493/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245864 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	bc16d44b73	fixes astobj2 unlinking of multiple objects when OBJ_MULTIPLE was disabled When OBJ_MULTIPLE was off but OBJ_UNLINK was on, all the items in a bucket were being unlinked instead of just the first match. This fixes that. Review: https://reviewboard.asterisk.org/r/490/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d4c861c61b	Oops git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	367f805919	Define a small set of constant return values git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244728 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	36bbf8f902	fixes crash during T.38 negotiation caused by invalid or missing FaxMaxDatagram field AST-2010-001 (closes issue #16634) Reported by: krn (closes issue #16724) Reported by: barthpbx (closes issue #16517) Reported by: bklang (closes issue #16485) Reported by: elsto git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244443 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	a170cd28e0	Add new option to asterisk.conf (lockconfdir) to protect conf dir during reloads (closes issue #16358) Reported by: raarts Patches: lockconfdir.diff uploaded by raarts (license 937) modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	d16b89be17	RFC compliant uri and display-name encode/decode 1. URI Encoding This patch changes ast_uri_encode()'s behavior when doreserved is enabled. Previously when doreserved was enabled only a small set of reserved characters were encoded. This set was comprised primarily of the reserved characters defined in RFC3261 section 25.1, but contained other characters as well. Rather than only escaping the reserved set, doreserved now escapes all characters not within the unreserved set as defined by RFC 3261 and RFC 2396. Also, the 'doreserved' variable has been renamed to 'do_special_char' in attempts to avoid confusion. When doreserve is not enabled, the previous logic of only encoding the characters <= 0X1F and > 0X7f remains, except for the '%' character, which must always be encoded as it signifies a HEX escaped character during the decode process. 2. URI Decoding: Break up URI before decode. In chan_sip.c ast_uri_decode is called on the entire URI instead of it's individual parts after it is parsed. This is not good as ast_uri_decode can introduce special characters back into the URI which can mess up parsing. This patch resolves this by not decoding a URI until parsing is completely done. There are many instances where we check to see if pedantic checking is enabled before we decode a URI. In these cases a new macro, SIP_PEDANTIC_DECODE, is used on the individual parsed segments of the URI rather than constantly putting if (pedantic) { decode() } checks everywhere in the code. In the areas where ast_uri_decode is not dependent upon pedantic checking this macro is not used, but decoding is still moved to each individual part of the URI. The only behavior that should change from this patch is the time at which decoding occurs. Since I had to look over every place URI parsing occurs to create this patch, I found several places where we use duplicate code for parsing. To consolidate the code, those areas have updated to use the parse_uri() function where possible. 3. SIP display-name decoding according to RFC3261 section 25. To properly decode the display-name portion of a FROM header, chan_sip's get_calleridname() function required a complete re-write. More information about this change can be found in the comments at the beginning of this function. 4. Unit Tests. Unit tests for ast_uri_encode, ast_uri_decode, and get_calleridname() have been written. This involved the addition of the test_utils.c file for testing the utils api. (closes issue #16299) Reported by: wdoekes Patches: astsvn-16299-get_calleridname.diff uploaded by wdoekes (license 717) get_calleridname_rewrite.diff uploaded by dvossel (license 671) Tested by: wdoekes, dvossel, Nick_Lewis Review: https://reviewboard.asterisk.org/r/469/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	7c61a7105f	Change api for pbx_builtin_setvar to actually return error code if a function can't be written to. This patch removes code that was duplicated from pbx.c to manager.c in order to prevent API change in released versions of Asterisk. There are propably also other places that would benefit from reading the return code and react if a function returns error codes on writing a value into it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242919 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	bc9f02a60d	Merged revisions 242520 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r242520 \| tilghman \| 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010) \| 8 lines Only rebuild bison and flex source files on demand, if bison and flex are detected by the configure script. Changed after discussion on the -dev list about possible unnecessary build failures, due to checkouts/untars causing these special source files to possibly be newer than their resulting C files. This should additionally ensure that nobody need learn about extra Makefile arguments to ensure the proper files get rebuilt when changes are made to these special source files. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	49bf540c71	Create iterative method for querying SRV results, and use that for finding AGI servers. (closes issue #14775) Reported by: _brent_ Patches: 20091215__issue14775.diff.txt uploaded by tilghman (license 14) hagi-5.patch uploaded by brent (license 388) Tested by: _brent_ Reviewboard: https://reviewboard.asterisk.org/r/378/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241188 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	568c057c4c	Extend max call limit duration from 24.8 days to 292+ million years. If the limit was set past MAX_INT upon answering, the call was immediately hung up due to overflow from the return of ast_tvdiff_ms (in ast_check_hangup). The time calculation functions ast_tvdiff_sec and ast_tvdiff_ms have been changed to return an int64_t to prevent overflow. Also the reporter suggested adding a message indicating the reason for the call hanging up. Given that the new limit is so much higher, the message (which would only really be useful in the overflow scenario) has been made a debug message only. (closes issue #16006) Reported by: viraptor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	ddad718f8e	Note where empty lines should reside in commit messages. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e8a6d2995e	Add pickup event to AMI. Also, fix AMI documentation. (closes issue #16431) Reported by: syspert Patches: 20100112__issue16431.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240421 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	ecbe7eff7a	Add the TESTTIME() dialplan function, which permits testing GotoIfTime. Specifically, by setting TESTTIME() to a particular date and time, you can test whether a dialplan correctly branches as was intended. This was developed after recent questions on the -users list on how to test their holiday dialplan logic. (closes issue #16464) Reported by: tilghman Patches: 20100112__issue16464.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/458/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	bd2c63a59d	Adding Tilghman's documentation from asterisk-dev to the actual file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239389 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	bf06747778	fixes AUDIOHOOK_INHERIT regression During the process of removing an audiohook from one channel and attaching it to another the audiohook's status is updated to DONE and then back to whatever it was previously. Typically updating the status after setting it to DONE is not a good idea because DONE can trigger unrecoverable audiohook destruction events... because of this a conditional check was added to audiohook_update_status to explicitly prevent the audiohook from ever changing after being set to DONE. It was this check that prevented audiohook inherit from work properly though. Now ast_audiohook_move_by_source is treated as a special exception, as the audiohook must be returned to its previous status after attaching it to the new channel. This is only a safe operation because the audiohook's lock is held the entire time, otherwise this could cause trouble. (closes issue #16522) Reported by: corruptor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	b70bc21627	fixes test.c compile issue when TEST_FRAMEWORK is not enabled The ast_test_status_update() function is defined in test.h. When TEST_FRAMEWORK is not enabled a macro is defined as a no-op place holder for this function. The macro did not contain the correct number of arguments. This caused a compile error. Much thanks to wdoekes for reporting the issue and supplying the patch! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	386b847075	Merged revisions 237405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237405 \| tilghman \| 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010) \| 16 lines Add a flag to disable the Background behavior, for AGI users. This is in a section of code that relates to two other issues, namely issue #14011 and issue #14940), one of which was the behavior of Background when called with a context argument that matched the current context. This fix broke FreePBX, however, in a post-Dial situation. Needless to say, this is an extremely difficult collision of several different issues. While the use of an exception flag is ugly, fixing all of the issues linked is rather difficult (although if someone would like to propose a better solution, we're happy to entertain that suggestion). (closes issue #16434) Reported by: rickead2000 Patches: 20091217__issue16434.diff.txt uploaded by tilghman (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: rickead2000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	2706de850a	Merged revisions 236585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236585 \| seanbright \| 2009-12-28 10:12:08 -0500 (Mon, 28 Dec 2009) \| 7 lines Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces. There was conditional code (based on build platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that was removed since it is fixed in newer versions of Solaris/OpenSolaris, but I am still running into it on Solaris 10 x86 so add a configure-time check for it. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236613 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	ffd9d82472	Allow test_heap.c to compile when AST_DEVMODE is true, but TEST_FRAMEWORK is false git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236185 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	73cb2d507b	Unit Test Framework API The Unit Test Framework is a new API that manages registration and execution of unit tests in Asterisk with the purpose of verifying the operation of C functions. The Framework consists of a single test manager accompanied by a list of registered test functions defined within the code. A test is defined, registered, and unregistered from the framework using a set of macros which allow the test code to only be compiled within asterisk when the TEST_FRAMEWORK flag is enabled in menuselect. This allows the test code to exist in the same file as the C functions it intends to verify. Registered tests may be viewed and executed via a set of new CLI commands. CLI commands are also present for generating and exporting test results into xml and txt formats. For more information and use cases please refer to the documentation provided at the beginning of the test.h file. Review: https://reviewboard.asterisk.org/r/447/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ef9be94b35	Change all refererences to 1.6.3 to be 1.8, since that will be the next feature release git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	cf7b67d9d3	Merged revisions 235635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235635 \| jpeeler \| 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009) \| 48 lines Correct CDR dispositions for BUSY/FAILED This patch is simple in that it reorders the disposition defines so that the fix for issue 12946 works properly (the default CDR disposition was changed to AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all CDR records are written. The side effects of CDR changes are scary, so I'm documenting the test cases performed to attempt to catch any regressions. The following tests were all performed using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls B (busy) Hangup C Hangup A (Both SIP and features) A calls B A blind transfers to C Hangup C (Both SIP and features) A calls B A attended transfers to C Hangup C A calls B A attended transfers to C (SIP) C blind transfers to A (features) Hangup A All of the test scenario CDRs matched. The following tests were performed just with the patch to ensure proper operation (with unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w) (closes issue #16180) Reported by: aatef Patches: bug16180.patch uploaded by jpeeler (license 325) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235660 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	6b34563778	Add auth_policy option to jabber.conf for auto user registration. The option is global and currently the acceptable values as noted in the sample config are accept or deny. (closes issue #15228) Reported by: lp0 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235342 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d4894b3d25	Is it Friday yet? git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	26daf50863	Add applications JabberJoin, JabberLeave, JabberSendGroup for XMPP groupchat (closes issue #14352) Reported by: fiddur Patches: trunk-14352-2.diff uploaded by phsultan (license 73) Tested by: fiddur git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	cfd17ef0a6	Move implementation of closefrom(3) from app.c to strcompat.c git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233358 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	aa9ec67f97	OS X does not define MSG_NOSIGNAL, but it does have a socket option SO_NOSIGPIPE. (closes issue #16178) Reported by: oej git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	7e0a2db236	Fix multiple issues with musiconhold, which led to classes not getting destroyed properly. * Classes are now tracked past removal from the core container, and module removal is actively prevented until all references are freed. * A hanging reference stored in the channel has been removed. This could have caused a mismatch and the music state not properly cleared, if two or more reloads occurred between MOH being stopped and MOH being restarted. * In certain circumstances, duplicate classes were possible. * A race existed at reload time between a process being killed and the thread responsible for reading from the related pipe respawning that process. * Several reference counts have also been corrected. At least one could have caused deleted classes to stick around forever, consuming resources. This originally manifested as MOH external processes that were not killed at reload time. (closes issue #16279, closes issue #16207) Reported by: parisioa, dcabot Patches: 20091202__issue16279__2.diff.txt uploaded by tilghman (license 14) Tested by: parisioa, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232660 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f46840c107	So apparently, some platforms don't have ffsll(3). The manpage lies; it says that the function is in POSIX, but that's only for ffs(3), not ffsll(3). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f59fe83c56	More 32->64 bit codec conversions. In the process of swapping ULAW to a place in the extended codec space, we found several unhandled cases, where a 32-bit integer was still being used to handle a codec field. Most of these have been fixed with this commit, although there is at least one case (codec_dahdi) which depends upon outside headers to be altered before a conversion can be made. (Fixes AST-278, SWP-459) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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2095 Commits (374e4e9d726ef6084180fdee7741f22b1efe60ce)