2017 Commits (1.6.2)

Author	SHA1	Message	Date
Mark Michelson	af727635c5	Merged revisions 264452 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r264452 | mmichelson | 2010-05-19 16:29:08 -0500 (Wed, 19 May 2010) | 86 lines Fix transcode_via_sln option with SIP calls and improve PLC usage. From reviewboard: The problem here is a bit complex, so try to bear with me... It was noticed by a Digium customer that generic PLC (as configured in codecs.conf) did not appear to actually be having any sort of benefit when packet loss was introduced on an RTP stream. I reproduced this issue myself by streaming a file across an RTP stream and dropping approx. 5% of the RTP packets. I saw no real difference between when PLC was enabled or disabled when using wireshark to analyze the RTP streams. After analyzing what was going on, it became clear that one of the problems faced was that when running my tests, the translation paths were being set up in such a way that PLC could not possibly work as expected. To illustrate, if packets are lost on channel A's read stream, then we expect that PLC will be applied to channel B's write stream. The problem is that generic PLC can only be done when there is a translation path that moves from some codec to SLINEAR. When I would run my tests, I found that every single time, read and write translation paths would be set up on channel A instead of channel B. There appeared to be no real way to predict which channel the translation paths would be set up on. This is where Kevin swooped in to let me know about the transcode_via_sln option in asterisk.conf. It is supposed to work by placing a read translation path on both channels from the channel's rawreadformat to SLINEAR. It also will place a write translation path on both channels from SLINEAR to the channel's rawwriteformat. Using this option allows one to predictably set up translation paths on all channels. There are two problems with this, though. First and foremost, the transcode_via_sln option did not appear to be working properly when I was placing a SIP call between two endpoints which did not share any common formats. Second, even if this option were to work, for PLC to be applied, there had to be a write translation path that would go from some format to SLINEAR. It would not work properly if the starting format of translation was SLINEAR. The one-line change presented in this review request in chan_sip.c fixed the first issue for me. The problem was that in sip_request_call, the jointcapability of the outbound channel was being set to the format passed to sip_request_call. This is nativeformats of the inbound channel. Because of this, when ast_channel_make_compatible was called by app_dial, both channels already had compatibly read and write formats. Thus, no translation path was set up at the time. My change is to set the jointcapability of the sip_pvt created during sip_request_call to the intersection of the inbound channel's nativeformats and the configured peer capability that we determined during the earlier call to create_addr. Doing this got the translation paths set up as expected when using transcode_via_sln. The changes presented in channel.c fixed the second issue for me. First and foremost, when Asterisk is started, we'll read codecs.conf to see the value of the genericplc option. If this option is set, and ast_write is called for a frame with no data, then we will attempt to fill in the missing samples for the frame. The implementation uses a channel datastore for maintaining the PLC state and for creating a buffer to store PLC samples in. Even when we receive a frame with data, we'll call plc_rx so that the PLC state will have knowledge of the previous voice frame, which it can use as a basis for when it comes time to actually do a PLC fill-in. So, reviewers, now I ask for your help. First off, there's the one line change in chan_sip that I have put in. Is it right? By my logic it seems correct, but I'm sure someone can tell me why it is not going to work. This is probably the change I'm least concerned about, though. What concerns me much more is the set of changes in channel.c. First off, am I even doing it right? When I run tests, I can clearly see that when PLC is activated, I see a significant increase in RTP traffic where I would expect it to be. However, in my humble opinion, the audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to me than when no PLC is used at all. I need someone to review the logic I have used to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm sure someone can point out somewhere where I've done something incorrectly. As I was writing this review request up, I decided to give the code a test run under valgrind, and I find that for some reason, calls to plc_rx are causing some invalid reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around a bit to see why that is the case. If it's obvious to someone reviewing, speak up! Finally, I have one other proposal that is not reflected in my code review. Since without transcode_via_sln set, one cannot predict or control where a translation path will be up, it seems to me that the current practice of using PLC only when transcoding to SLINEAR is not useful. I recommend that once it has been determined that the method used in this code review is correct and works as expected, then the code in translate.c that invokes PLC should be removed. Review: https://reviewboard.asterisk.org/r/622/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f134c6effe	Merged revisions 264249 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r264249 | tilghman | 2010-05-19 12:48:31 -0500 (Wed, 19 May 2010) | 24 lines Merged revisions 264248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19 May 2010) | 17 lines Internal timing is now on by default, if you're using DAHDI 2.3 or above. The reason for ensuring DAHDI 2.3 or above is that this version ensures that a timer is always available, whereas in previous versions, it was possible for DAHDI to be loaded, but have no drivers to actually generate timing. If internal_timing was turned on in this circumstance, a complete lack of audio would result. This is the reason why internal_timing was not on by default. However, now that DAHDI ensures the availability of a timer, there is no reason for this setting to be off (and in fact, it solves a great many initial user problems). (closes issue #15932) Reported by: dimas Patches: 20100519__issue15932.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264250 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	2a3a1fe3cf	Merged revisions 262513 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r262513 | tilghman | 2010-05-11 16:25:05 -0500 (Tue, 11 May 2010) | 7 lines Move cause 200 to cause 26, as specified in Q.850. Also cleanup the formatting and add a few more that seem like good candidates. (closes issue #16157) Reported by: wimpy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@262516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	15e31273f5	Merged revisions 262102 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r262102 | tilghman | 2010-05-08 21:14:04 -0500 (Sat, 08 May 2010) | 5 lines Cleanup a bit more by getting rid of useless version defines. Also make library detection use passed CFLAGS. (closes issue #17309) Reported by: stuarth ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@262105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c3cee12c21	Merged revisions 261913 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r261913 | tilghman | 2010-05-07 15:35:17 -0500 (Fri, 07 May 2010) | 14 lines Use the detected pthread building flags in every place, instead of hardcoding -lpthread. We nicely detect the right flags on each system for building Asterisk with pthreads, then ignore it for every other build option that requires us to build with pthreads. This caused some items to return a false negative. Also cleanup some minor naming issues that caused "library library" redundancy in the output. (closes issue #17303) Reported by: stuarth Patches: 20100507__issue17303.diff.txt uploaded by tilghman (license 14) Tested by: stuarth ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@261916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	55993d7b13	Merged revisions 254450 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r254450 | kpfleming | 2010-03-25 10:27:31 -0500 (Thu, 25 Mar 2010) | 49 lines Improve handling of T.38 re-INVITEs that arrive before a T.38-capable application is executing on a channel. This patch addresses an issue found during working with end-users using res_fax. If an incoming call is answered in the dialplan, or jumps to the 'fax' extension due to reception of a CNG tone (with faxdetect enabled), and then the remote endpoint sends a T.38 re-INVITE, it is possible for the channel's T.38 state to be 'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately, even if the application wants to use T.38, it can't respond to the peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent originally has been lost, and the application needs the content of that frame to be able to formulate a reply. This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS, AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip will re-send the original control frame (with AST_T38_REQUEST_NEGOTIATE as the request type), and the application can respond as normal. If this occurs within the five second timeout in chan_sip, the automatic cancellation of the peer reinvite will be stopped, and the application will 'own' the negotiation process from that point onwards. This also improves the code path in chan_sip to allow sip_indicate(), when called for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero response, which should have been in place before since the control frame *can* fail to be processed properly. It also modifies ast_indicate() to return whatever result the channel driver returned for this control frame, rather than converting all non-zero results into '-1'. Finally, the new request type intentionally returns a positive value, so that an application that sends AST_T38_REQUEST_PARMS can know for certain whether the channel driver accepted it and will be replying with a control frame of its own, or whether it was ignored (if the sip_indicate()/ast_indicate() path had properly supported failure responses before, this would not be necessary). This patch also modifies res_fax to take advantage of the new request. In addition, this patch makes sip_t38_abort() actually lock the private structure before doing its work... bad programmer, no donut. This patch also enhances chan_sip's 'faxdetect' support to allow triggering on T.38 re-INVITEs received as well as CNG tone detection. Review: https://reviewboard.asterisk.org/r/556/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@260884 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	4b4569810a	Merged revisions 260050 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r260050 | dvossel | 2010-04-29 10:33:27 -0500 (Thu, 29 Apr 2010) | 21 lines Merged revisions 260049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010) | 14 lines Fixes crash in audiohook_write_list The middle_frame in the audiohook_write_list function was being freed if a audiohook manipulator returned a failure. This is incorrect logic. This patch resolves this and adds detailed descriptions of how this function should work and why manipulator failures must be ignored. (closes issue #17052) Reported by: dvossel Tested by: dvossel (closes issue #16196) Reported by: atis Review: https://reviewboard.asterisk.org/r/623/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@260051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e862c28407	Merged revisions 257560 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r257560 | tilghman | 2010-04-15 16:26:19 -0500 (Thu, 15 Apr 2010) | 13 lines Merged revisions 257544 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010) | 6 lines Allow application options with arguments to contain parentheses, through a variety of escaping techniques. Fixes SWP-1194 (ABE-2143). Review: https://reviewboard.asterisk.org/r/604/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@257597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d680d86948	Merged revisions 256370 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r256370 | tilghman | 2010-04-06 14:28:42 -0500 (Tue, 06 Apr 2010) | 2 lines Mac OS X does not support comparing a mutex to its initializer. Create a test for this. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@256373 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	52715df18f	Merged revisions 255796 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r255796 | tilghman | 2010-04-01 13:16:37 -0500 (Thu, 01 Apr 2010) | 7 lines Fix DEBUG_THREADS build on Darwin. (closes issue #16828) Reported by: oej Patches: 20100331__issue16828.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@255816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	3aed281c75	Fix DEBUG_THREADS issue with out-of-tree modules. ... Take 2, without ABI breakage this time. Review: https://reviewboard.asterisk.org/r/588/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@254770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	1c36715ade	Merged revisions 254553 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r254553 | mmichelson | 2010-03-25 12:42:36 -0500 (Thu, 25 Mar 2010) | 11 lines Merged revisions 254552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu, 25 Mar 2010) | 5 lines Add doxygen for acl.h Review: https://reviewboard.asterisk.org/r/528 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@254556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	dbf0243679	Revert API change in release branches ... This re-renames ast_rtp_update_source to ast_rtp_new_source git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@253158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	6bfb1fca95	Merged revisions 252089 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines 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/branches/1.6.2@252137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	07a05534e0	Merged revisions 249893 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 | dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines 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/branches/1.6.2@249895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	b8fd3f54b2	Merged revisions 249405 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r249405 | tilghman | 2010-02-28 01:10:22 -0600 (Sun, 28 Feb 2010) | 2 lines Properly document voicemail API documents. Also fix a crash reported via the -dev list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@249407 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	a03d04345a	Merged revisions 247335 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r247335 | mmichelson | 2010-02-17 15:22:40 -0600 (Wed, 17 Feb 2010) | 20 lines 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/branches/1.6.2@247337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	872c5f5412	Merged revisions 246985 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r246985 | mmichelson | 2010-02-16 15:15:38 -0600 (Tue, 16 Feb 2010) | 3 lines Add some clarifying documentation to the ast_str_set and ast_str_append functions. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@246989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e2e66d446c	Merged revisions 246030 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r246030 | tilghman | 2010-02-10 10:01:28 -0600 (Wed, 10 Feb 2010) | 12 lines 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/branches/1.6.2@246199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	e8162c4245	Merged revisions 244443 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r244443 | dvossel | 2010-02-02 16:27:23 -0600 (Tue, 02 Feb 2010) | 18 lines 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/branches/1.6.2@244445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c2746e94a7	Merged revisions 242521 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r242521 | tilghman | 2010-01-24 00:40:31 -0600 (Sun, 24 Jan 2010) | 15 lines 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/branches/1.6.2@242522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	46c684eed9	Merged revisions 238635 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r238635 | dvossel | 2010-01-08 13:39:30 -0600 (Fri, 08 Jan 2010) | 22 lines 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/branches/1.6.2@238637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	5bc3fe1baa	Merged revisions 209400 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r209400 | kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3 lines Define side-effect-safe MIN and MAX macros and remove duplicate definitions from various files. (closes issue #16251) Reported by: asgaroth ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@238499 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	21ddcd4709	Merged revisions 237406 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r237406 | tilghman | 2010-01-04 12:28:28 -0600 (Mon, 04 Jan 2010) | 23 lines 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/branches/1.6.2@237409 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	d05977c54f	Merged revisions 236613 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r236613 | seanbright | 2009-12-28 10:22:54 -0500 (Mon, 28 Dec 2009) | 14 lines 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/branches/1.6.2@236635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	4f7c2bdf62	Merged revisions 235660 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r235660 | jpeeler | 2009-12-18 16:51:37 -0600 (Fri, 18 Dec 2009) | 55 lines 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/branches/1.6.2@235665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	98ecc431c9	Merged revisions 231692 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r231692 | kpfleming | 2009-11-30 15:47:42 -0600 (Mon, 30 Nov 2009) | 22 lines Another round of UDPTL stack fixes/improvements: 1) Allow users of UDPTL stack to associate a character-string tag with a UDPTL session, so that log/error/debug messages generated by the UDPTL stack can be 'connected' to the endpoint that caused them to be generated. 2) Improve comments (and process) of calculating the far end's maximum IFP size when redundancy mode is in use for error correction. 3) When an IFP larger than the calculated 'far max IFP' size is presented for writing, truncate it rather than putting in the buffer and allowing the buffer to overflow; this will cause the ends to retrain to a lower bit rate that produces IFPs of an appropriate size if possible, and if not possible, the FAX transfer will fail completely. In these cases, it is due to the one endpoint supplying a T38FaxMaxDatagram value that is improperly calculated and is too low to be of use; we have configuration options available to override this behavior. 4) Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no longer needed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@231696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	6f122d97ae	Merged revisions 231688 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r231688 | mnicholson | 2009-11-30 15:31:55 -0600 (Mon, 30 Nov 2009) | 15 lines Merged revisions 231614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@231689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f54a6f1e07	Merged revisions 230583 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r230583 | dvossel | 2009-11-20 11:26:20 -0600 (Fri, 20 Nov 2009) | 6 lines audiohook signal trigger on every status change (issue #14618) Review: https://reviewboard.asterisk.org/r/434/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@230585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	e137b23cb5	Merged revisions 228858 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r228858 | mnicholson | 2009-11-09 08:37:07 -0600 (Mon, 09 Nov 2009) | 15 lines Merged revisions 228827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon, 09 Nov 2009) | 8 lines Perform limited bounds checking when destroying ast_mutex_t structures to make sure we don't try to use negative indices. (closes issue #15588) Reported by: zerohalo Patches: 20090820__issue15588.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@228859 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	56f95a4d9c	Merged revisions 226305 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r226305 | tilghman | 2009-10-28 13:04:05 -0500 (Wed, 28 Oct 2009) | 9 lines Merged revisions 226304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28 Oct 2009) | 2 lines Fix documentation (pointed out by TheDavidFactor on #-dev) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@226308 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	12fb1e8bde	Merged revisions 225445 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r225445 | dvossel | 2009-10-22 14:55:51 -0500 (Thu, 22 Oct 2009) | 50 lines SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes. What this patch fixes 1.Moves sip TCP/TLS connection setup into the TCP helper thread: Connection setup takes awhile and before this it was being done while holding the monitor lock. 2.Moves TCP/TLS writing to the TCP helper thread: Through the use of a packet queue and an alert pipe, the TCP helper thread can now be woken up to write data as well as read data. 3.Locking error: sip_xmit returned an XMIT_ERROR without giving up the tcptls_session lock. This lock has been completely removed from sip_xmit and placed in the new sip_tcptls_write() function. 4.Memory leak: When creating a tcptls_client the tls_cfg was alloced but never freed unless the tcptls_session failed to start. Now the session_args for a sip client are an ao2 object which frees the tls_cfg on destruction. 5.Pointer to stack variable: During sip_prepare_socket the creation of a client's ast_tcptls_session_args was done on the stack and stored as a pointer in the newly created tcptls_session. Depending on the events that followed, there was a slight possibility that pointer could have been accessed after the stack returned. Given the new changes, it is always accessed after the stack returns which is why I found it. Notable code changes 1.I broke tcptls.c's ast_tcptls_client_start() function into two functions. One for creating and allocating the new tcptls_session, and a separate one for starting and handling the new connection. This allowed me to create the tcptls_session, launch the helper thread, and then establish the connection within the helper thread. 2.Writes to a tcptls_session are now done within the helper thread. This is done by using an alert pipe to wake up the thread if new data needs to be sent. The thread's sip_threadinfo object contains the alert pipe as well as the packet queue. 3.Since the threadinfo object contains the alert pipe, it must now be accessed outside of the helper thread for every write (queuing of a packet). For easy lookup, I moved the threadinfo objects from a linked list to an ao2_container. (closes issue #13136) Reported by: pabelanger Tested by: dvossel, whys (closes issue #15894) Reported by: dvossel Tested by: dvossel Review: https://reviewboard.asterisk.org/r/380/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@225489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	c6a9fbf2c9	Merged revisions 225360 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009) | 11 lines Merged revisions 225105 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines Fix documentation for ast_softhangup() and correct the misuse thereof. (closes issue #16103) Reported by: majorbloodnok ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@225363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	bfd8af8cc8	Merged revisions 222880 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r222880 | russell | 2009-10-08 14:52:03 -0500 (Thu, 08 Oct 2009) | 51 lines Merged revisions 222878 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines Make filestream frame handling safer by isolating frames before returning them. This patch is related to a number of issues on the bug tracker that show crashes related to freeing frames that came from a filestream. A number of fixes have been made over time while trying to figure out these problems, but there re still people seeing the crash. (Note that some of these bug reports include information about other problems. I am specifically addressing the filestream frame crash here.) I'm still not clear on what the exact problem is. However, what is _very_ clear is that we have seen quite a few problems over time related to unexpected behavior when we try to use embedded frames as an optimization. In some cases, this optimization doesn't really provide much due to improvements made in other areas. In this case, the patch modifies filestream handling such that the embedded frame will not be returned. ast_frisolate() is used to ensure that we end up with a completely mallocd frame. In reality, though, we will not actually have to malloc every time. For filestreams, the frame will almost always be allocated and freed in the same thread. That means that the thread local frame cache will be used. So, going this route doesn't hurt. With this patch in place, some people have reported success in not seeing the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2) Tested by: aragon, russell (closes issue #15817) Reported by: zerohalo Tested by: zerohalo (closes issue #15845) Reported by: marhbere Review: https://reviewboard.asterisk.org/r/386/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@222883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	d9a817f431	Merged revisions 222873 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r222873 | dvossel | 2009-10-08 14:35:30 -0500 (Thu, 08 Oct 2009) | 6 lines fixes an ast_netsock_list memory leak. ABE-1998 Review: https://reviewboard.asterisk.org/r/395/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@222874 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	6410aa2418	Merged revisions 222176 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines Fix ao2_iterator API to hold references to containers being iterated. See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@222187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	9a61ba3e28	Merged revisions 221266 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines Merged revisions 221086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@221304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	4ddccae801	Remove optional_api from 1.6.2 branch, since it is not currently working. ... This is a blocking issue for the 1.6.2 release. (closes issue #15914) Reported by: mbeckwell Branch: http://svn.digium.com/svn/asterisk/team/tilghman/optional_api_162 Tested by: mbeckwell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@221042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	b1848ad0fc	Merged revisions 220586 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r220586 | tilghman | 2009-09-26 10:10:28 -0500 (Sat, 26 Sep 2009) | 2 lines Allow AES to compile, when OpenSSL is not present. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@220589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	5eb7d1c6b1	Merged revisions 219139 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r219139 | mnicholson | 2009-09-17 10:18:01 -0500 (Thu, 17 Sep 2009) | 17 lines Merged revisions 219136 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down. This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h. (closes issue #15316) Reported by: vmarrone Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/362/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@219194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	571890c856	Merged revisions 217638 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r217638 | tilghman | 2009-09-10 13:17:14 -0500 (Thu, 10 Sep 2009) | 4 lines Verify support for wide ODBC character types before using them. (closes issue #15870) Reported by: nic_bellamy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@217647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	2b2b717d71	Merged revisions 217074 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r217074 | kpfleming | 2009-09-08 11:37:28 -0500 (Tue, 08 Sep 2009) | 9 lines Ensure that the default autoconf CFLAGS are not used. A recent change to the configure script that allows the user to specify CFLAGS and/or LDFLAGS to the script had the unfortunate side effect of letting autoconf's default CFLAGS (-g -O2) feed in to the rest of the build system, thereby overriding the DONT_OPTIMIZE setting in menuselect. That problem is now corrected. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@217077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	8a9ab86721	Merged revisions 216551 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r216551 | tilghman | 2009-09-04 12:50:21 -0500 (Fri, 04 Sep 2009) | 2 lines Fix trunk breakage. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@216553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	4f750dac83	Merged revisions 216437 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r216437 | mvanbaak | 2009-09-04 16:00:38 +0200 (Fri, 04 Sep 2009) | 2 lines make sure canlog is set so we can compile with DEBUG_THREADS enabled on OpenBSD ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@216469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	574c8c2144	Merged revisions 215955 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r215955 | dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009) | 6 lines Merge code associated with AST-2009-006 (closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@216007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	5372378abb	Merged revisions 214819 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r214819 | tilghman | 2009-08-30 01:43:04 -0500 (Sun, 30 Aug 2009) | 4 lines If lua is detected with the lua5.1 prefix (or not), adjust the include path accordingly. Based upon feedback to a release announcement on the -users list. See http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@214822 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	7838215111	Merged revisions 214696 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r214696 | kpfleming | 2009-08-28 15:01:21 -0500 (Fri, 28 Aug 2009) | 9 lines Ensure that CFLAGS and/or LDFLAGS provided to configure script are preserved. Cross-compilation environments want to provide 'defaults' for compiler and linker options, and frequently do this by specifying CFLAGS and LDFLAGS in the environment or as command-line arguments to the configure script. This patch modifies the configure script and Makefile to preserve these settings and ensure they are used in the build process. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@214700 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	7330132ee7	Merged revisions 214650 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r214650 | mmichelson | 2009-08-28 13:41:23 -0500 (Fri, 28 Aug 2009) | 3 lines Fix some incorrect documentation of sched_thread functions. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@214653 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	23a70136b7	Merged revisions 214518 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r214518 | tilghman | 2009-08-27 16:46:46 -0500 (Thu, 27 Aug 2009) | 14 lines Merged revisions 214517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27 Aug 2009) | 7 lines Use autoconf to detect libcurl, as this enables cross-compilation checks, something we didn't allow before. (closes issue #15714) Reported by: pprindeville Patches: 20090813__issue15714.diff.txt uploaded by tilghman (license 14) Tested by: pprindeville ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@214521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ef5ae5be67	Merged revisions 214466 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r214466 | tilghman | 2009-08-27 12:28:01 -0500 (Thu, 27 Aug 2009) | 9 lines Merged revisions 214436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27 Aug 2009) | 2 lines One more build system change, to make the descriptions look better, if we have better information. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@214504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	10e51e82f1	Merged revisions 214360 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r214360 | tilghman | 2009-08-27 11:12:03 -0500 (Thu, 27 Aug 2009) | 10 lines Merged revisions 214357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27 Aug 2009) | 3 lines Make autoheader descriptions render correctly in our autoconfig.h file. (Figured out while working with issue #14906) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@214363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	aa1b25b24d	Merged revisions 214152 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r214152 | tilghman | 2009-08-25 17:39:51 -0500 (Tue, 25 Aug 2009) | 4 lines Not all versions of gnu-linux use glibc, which contains iconv. Some (especially embedded systems) don't have iconv at all. (closes issue #15169) Reported by: pprindeville ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@214155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	7bc6aaf989	Merged revisions 213560 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r213560 | tilghman | 2009-08-21 11:53:52 -0500 (Fri, 21 Aug 2009) | 14 lines Merged revisions 213559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213559 | tilghman | 2009-08-21 11:52:53 -0500 (Fri, 21 Aug 2009) | 7 lines Permit DEBUG_FD_LEAKS to be used with C++ source files. (closes issue #15698) Reported by: slavon Patches: 20090817__issue15698.diff.txt uploaded by tilghman (license 14) Tested by: slavon, tilghman ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@213563 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	fe6107aba0	Merged revisions 210908 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 | tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines Allow Gosub to recognize quote delimiters without consuming them. (closes issue #15557) Reported by: rain Patches: 20090723__issue15557.diff.txt uploaded by tilghman (license 14) Tested by: rain Review: https://reviewboard.asterisk.org/r/316/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@210911 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	503bb8de9c	Merged revisions 209554 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 | dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines Fixes numerous spelling errors. Patch submitted by alecdavis. (closes issue #15595) Reported by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@209594 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	4921462e88	Merged revisions 209098 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 | dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize" (closes issue #15571) Reported by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@209234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	b7e75a4a56	Merged revisions 208548 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 | kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8 lines Resolve a T.38 negotiation issue left over from the udptl-updates merge. The udptl-updates branch that was merged yesterday failed to properly send back T.38 SDP responses with the correct error correction mode, if the incoming SDP from the other end caused us to change error correction modes. This patch corrects that situation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@208551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	03d73280b3	Merged revisions 208464 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines Rework of T.38 negotiation and UDPTL API to address interoperability problems Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum *IFP* size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending *our* MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@208501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	b4ad95ecf3	Merged revisions 205696 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r205696 | kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16 lines Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover. Recent changes in T.38 negotiation in Asterisk caused these applications to not respond when the other endpoint initiated a switchover to T.38; this resulted in the T.38 switchover failing, and the FAX attempt to be made using an audio connection, instead of T.38 (which would usually cause the FAX to fail completely). This patch corrects this problem, and the applications will now correctly respond to the T.38 switchover request. In addition, the response will include the appopriate T.38 session parameters based on what the other end offered and what our end is capable of. (closes issue #14849) Reported by: afosorio ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	8a7c94dae5	Merged revisions 205600 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r205600 | dvossel | 2009-07-09 11:19:09 -0500 (Thu, 09 Jul 2009) | 9 lines Merged revisions 205599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) | 2 lines Changing ast_samp2tv to not use floating point. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	4088c54d14	Merged revisions 205479 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009) | 16 lines Merged revisions 205471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	a076c9af21	Merged revisions 205412 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009) | 12 lines Merged revisions 205409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines moving ast_devstate_to_extenstate to pbx.c from devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This change fixes a compile time error with chan_vpb as well. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	49ac045c97	Merged revisions 205216 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r205216 | dvossel | 2009-07-08 11:54:24 -0500 (Wed, 08 Jul 2009) | 17 lines Merged revisions 205215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) | 10 lines ast_samp2tv needs floating point for 16khz audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The .5 is currently stripped off because we don't calculate using floating points. This causes madness with 16khz audio. (issue ABE-1899) Review: https://reviewboard.asterisk.org/r/305/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	e97c471b2c	Merged revisions 205120 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines Move OpenSSL initialization to a single place, make library usage thread-safe. While doing some reading about OpenSSL, I noticed a couple of things that needed to be improved with our usage of OpenSSL. 1) We had initialization of the library done in multiple modules. This has now been moved to a core function that gets executed during Asterisk startup. We already link OpenSSL into the core for TCP/TLS functionality, so this was the most logical place to do it. 2) OpenSSL is not thread-safe by default. However, making it thread safe is very easy. We just have to provide a couple of callbacks. One callback returns a thread ID. The other handles locking. For more information, start with the "Is OpenSSL thread-safe?" question on the FAQ page of openssl.org. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	e5556be26b	Merged revisions 204710 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009) | 21 lines Merged revisions 204681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines Improved mapping of extension states from combined device states. This fixes a few issues with incorrect extension states and adds a cli command, core show device2extenstate, to display all possible state mappings. (closes issue #15413) Reported by: legart Patches: exten_helper.diff uploaded by dvossel (license 671) Tested by: dvossel, legart, amilcar Review: https://reviewboard.asterisk.org/r/301/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204711 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	e3e62b491b	Merged revisions 203702 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r203702 | russell | 2009-06-26 14:31:14 -0500 (Fri, 26 Jun 2009) | 5 lines Make invalid hints report Unavailable instead of Idle. (closes issue #14413) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	45d9ad47ca	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/1.6.2@203705 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	824737f6d7	Merged revisions 202410 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r202410 | dvossel | 2009-06-22 10:33:35 -0500 (Mon, 22 Jun 2009) | 5 lines attempting to load running modules Modules placed in the priority heap for loading were not properly removed from the linked list. This resulted in some modules attempting to load twice. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@202411 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	a66df3a4ab	Merged revisions 201262 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r201262 | kpfleming | 2009-06-17 07:04:17 -0500 (Wed, 17 Jun 2009) | 15 lines Merged revisions 201261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty. When the list to be appended is empty, and the list to be appended to is *not*, AST_LIST_APPEND_LIST would actually cause the target list to become broken, and no longer have a pointer to its last entry. This patch fixes the problem. (reported by Stanislaw Pitucha on the asterisk-dev mailing list) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@201265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	366795b84e	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/1.6.2@201097 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	08098c0968	Merged revisions 200764 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15 Jun 2009) | 11 lines Ensure that configure-script testing for compiler attributes actually works. The configure script tests for compiler attributes didn't actually enable enough warnings or provide a proper test harness to determine whether the compiler supports the attribute in question or not; this caused gcc 4.1 to report that it supports 'weakref', but it doesn't actually support it in the way that is needed for our optional API mechanism. The new configure script test will properly distinguish between full support and partial support for this attribute, among others. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@200765 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	5b2345c198	Merged revisions 199857 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400 (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed, 10 Jun 2009) | 2 lines __WORDSIZE is not available on all platforms, so use sizeof(void *) instead. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@199860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	443cb28322	Merged revisions 199743 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r199743 | dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009) | 11 lines module load priority This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized. The lower the value, the higher the priority. The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority on load. Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty. (closes issue #15191) Reported by: alecdavis Tested by: dvossel Review: https://reviewboard.asterisk.org/r/262/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@199744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	fb39ec3d2d	Merged revisions 199630 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400 (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) | 21 lines Increase the size of our thread stack on 64 bit processors. We were setting the stack size for each thread to 240KB regardless of architecture, which meant that in some scenarios we actually had less available stack space on 64 bit processors (pointers use 8 bytes instead of 4). So now we calculate the stack size we reserve based on the platform's __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128 bit -> 1008KB (that's right, we're ready for 128 bit processors) Patch typed by me but written by several members of #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes issue #14932) Reported by: jpiszcz Patches: 06052009_issue14932.patch uploaded by seanbright (license 71) Tested by: seanbright ........ r199628 | seanbright | 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the stack size calculation just introduced. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@199634 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	1b47d4acb8	Merged revisions 199298 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009) | 21 lines Merged revisions 199297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) | 14 lines Fixes issue with hints giving unexpected results. Hints with two or more devices that include ONHOLD gave unexpected results. (closes issue #15057) Reported by: p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel (license 671) pbx.c.1.4.patch uploaded by p (license 558) devicestate.c.trunk.patch uploaded by p (license 671) Tested by: p_lindheimer, dvossel Review: https://reviewboard.asterisk.org/r/254/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@199299 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	b4b61e5273	Merged revisions 199051 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun 2009) | 47 lines Merged revisions 199022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines Safely handle AMI connections/reload requests that occur during startup. During asterisk startup, a lock on the list of modules is obtained by the primary thread while each module is initialized. Issue 13778 pointed out a problem with this approach, however. Because the AMI is loaded before other modules, it is possible for a module reload to be issued by a connected client (via Action: Command), causing a deadlock. The resolution for 13778 was to move initialization of the manager to happen after the other modules had already been lodaded. While this fixed this particular issue, it caused a problem for users (like FreePBX) who call AMI scripts via an #exec in a configuration file (See issue 15189). The solution I have come up with is to defer any reload requests that come in until after the server is fully booted. When a call comes in to ast_module_reload (from wherever) before we are fully booted, the request is added to a queue of pending requests. Once we are done booting up, we then execute these deferred requests in turn. Note that I have tried to make this a bit more intelligent in that it will not queue up more than 1 request for the same module to be reloaded, and if a general reload request comes in ('module reload') the queue is flushed and we only issue a single deferred reload for the entire system. As for how this will impact existing installations - Before 13778, a reload issued before module initialization was completed would result in a deadlock. After 13778, you simply couldn't connect to the manager during startup (which causes problems with #exec-that-calls-AMI configuration files). I believe this is a good general purpose solution that won't negatively impact existing installations. (closes issue #15189) (closes issue #13778) Reported by: p_lindheimer Patches: 06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71) Tested by: p_lindheimer, seanbright Review: https://reviewboard.asterisk.org/r/272/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@199054 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	2b53e7117f	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/1.6.2@198886 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	2f929aeb79	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/1.6.2@198075 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	ee9fded518	Merged revisions 197606 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) | 22 lines Recorded merge of revisions 197588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines Allow for media to arrive from an alternate source when responding to a reinvite with 491. When we receive a SIP reinvite, it is possible that we may not be able to process the reinvite immediately since we have also sent a reinvite out ourselves. The problem is that whoever sent us the reinvite may have also sent a reinvite out to another party, and that reinvite may have succeeded. As a result, even though we are not going to accept the reinvite we just received, it is important for us to not have problems if we suddenly start receiving RTP from a new source. The fix for this is to grab the media source information from the SDP of the reinvite that we receive. This information is passed to the RTP layer so that it will know about the alternate source for media. Review: https://reviewboard.asterisk.org/r/252 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@197619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	652b9b510b	Merged revisions 197543 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r197543 | mmichelson | 2009-05-28 09:58:06 -0500 (Thu, 28 May 2009) | 27 lines Merged revisions 197537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines Add flags to chanspy audiohook so that audio stays in sync. There are two flags being added to the chanspy audiohook here. One is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that the read and write slinfactories on the audiohook do not skew beyond a certain tolerance. In addition, there is a new audiohook flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for a slinfactory to build up a substantial amount of audio before flushing it. For this particular issue, this means that the person spying on the call will hear the conversations in real time with very little delay in the audio. (closes issue #13745) Reported by: geoffs Patches: 13745.patch uploaded by mmichelson (license 60) Tested by: snblitz ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@197548 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	ee8c298cae	Merged revisions 197335 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r197335 | kpfleming | 2009-05-27 17:21:53 -0500 (Wed, 27 May 2009) | 3 lines Ensure that this header includes xmldoc.h, since it depends on it. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@197336 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	c1e7c6b1bf	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/1.6.2@197176 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	77cfa5b791	Merged revisions 196946 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r196946 | russell | 2009-05-26 17:40:34 -0500 (Tue, 26 May 2009) | 8 lines Update configure script to check for OSP toolkit 3.5.0. (closes issue #14988) Reported by: tzafrir Patches: configure.ac.diff uploaded by homesick (license 91) new_ast_check_osptk.m4 uploaded by homesick (license 91) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@196949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	0c3e56ea9a	Merged revisions 195882 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r195882 | mnicholson | 2009-05-21 10:33:55 -0500 (Thu, 21 May 2009) | 20 lines Merged revisions 195881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) | 13 lines This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases. This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected. This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times. (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes issue #14744) Reported by: deepesh ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@195883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	bb4849e44a	Merged revisions 195370 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r195370 | tilghman | 2009-05-18 15:52:33 -0500 (Mon, 18 May 2009) | 15 lines Recorded merge of revisions 195366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009) | 8 lines Add a similar dependency on SMDI for voicemail as already exists for ADSI. (closes issue #14846) Reported by: pj Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt uploaded by tilghman (license 14) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@195373 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	9e97695955	Merged revisions 193461 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r193461 | russell | 2009-05-09 06:33:09 -0500 (Sat, 09 May 2009) | 2 lines Minor documentation update for ast_event_queue(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@193462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	c2883a4eb3	Merged revisions 192357 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r192357 | kpfleming | 2009-05-05 15:18:21 +0200 (Tue, 05 May 2009) | 5 lines Correct some flaws in the memory accounting code for stringfields and ao2 objects Under some conditions, the memory allocation for stringfields and ao2 objects would not have supplied valid file/function names for MALLOC_DEBUG tracking, so this commit corrects that. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@192360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	726ceee0c3	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/1.6.2@192355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	6f6019c309	Merged revisions 192279 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r192279 | kpfleming | 2009-05-05 10:51:06 +0200 (Tue, 05 May 2009) | 5 lines Ensure that string pools allocated to hold stringfields are properly accounted in MALLOC_DEBUG mode This commit modifies the stringfield pool allocator to remember the 'owner' of the stringfield manager the pool is being allocated for, and ensures that pools allocated in the future when fields are populated are owned by that file/function. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@192298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	7c9986fea2	Merged revisions 192059 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r192059 | kpfleming | 2009-05-04 18:24:16 +0200 (Mon, 04 May 2009) | 5 lines Ensure that astobj2 memory allocations are properly accounted for when MALLOC_DEBUG is used This commit ensures that all astobj2 allocated objects are properly accounted for in MALLOC_DEBUG mode by passing down the file/function/line information from the module/function that actually called the astobj2 allocation function. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@192175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	015d00b612	Merged revisions 191367 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r191367 | tilghman | 2009-04-30 12:40:58 -0500 (Thu, 30 Apr 2009) | 3 lines Detect eaccess (or euidaccess) before using it. Reported by Andrew Lindh via the -dev list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@191370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	37ad95b96a	Merged revisions 190725 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r190725 | kpfleming | 2009-04-27 14:30:54 -0500 (Mon, 27 Apr 2009) | 13 lines Merged revisions 190721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr 2009) | 7 lines Fix 'inconsistent line endings' when autoconf 2.63 is used Attempt to make configure script regeneration 'safe' using autoconf 2.63, which embeds a bare CR into the script, thus making Subversion complain about inconsistent line endings This commit changes the MIME type of the configure script to be 'binary' thus making Subversion no longer inspect line endings, and as a bonus 'svn diff' will no longer try to generate diff output for it, which is not generally useful anyway. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@190732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	a8af53fe04	Merged revisions 190093 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r190093 | tilghman | 2009-04-22 16:38:15 -0500 (Wed, 22 Apr 2009) | 14 lines Merged revisions 190092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190092 | tilghman | 2009-04-22 16:35:03 -0500 (Wed, 22 Apr 2009) | 7 lines Detect availability of pthread_rwlock_timedwrlock() before using it. (closes issue #14930) Reported by: tilghman Patches: 20090420__bug14930.diff.txt uploaded by tilghman (license 14) Tested by: mvanbaak, tilghman ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@190096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	e8a4ceb8e5	Merged revisions 190057 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r190057 | jpeeler | 2009-04-22 16:15:55 -0500 (Wed, 22 Apr 2009) | 9 lines Fix building of chan_h323 with gcc-3.3 There seems to be a bug with old versions of g++ that doesn't allow a structure member to use the name list. Rename list member to group_list in ast_group_info and change the few places it is used. (closes issue #14790) Reported by: stuarth ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@190066 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	6d32c4c168	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/1.6.2@187601 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f063970ef2	Merged revisions 187483 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r187483 | tilghman | 2009-04-09 13:40:01 -0500 (Thu, 09 Apr 2009) | 15 lines Merged revisions 187428 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009) | 8 lines Race condition between ast_cli_command() and 'module unload' could cause a deadlock. Add lock timeouts to avoid this potential deadlock. (closes issue #14705) Reported by: jamessan Patches: 20090320__bug14705.diff.txt uploaded by tilghman (license 14) Tested by: jamessan ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@187487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	03c25bb848	Merged revisions 187302 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r187302 | tilghman | 2009-04-08 23:59:05 -0500 (Wed, 08 Apr 2009) | 14 lines Merged revisions 187300-187301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) | 3 lines Add debugging mode for diagnosing file descriptor leaks. (Related to issue #14625) ........ r187301 | tilghman | 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops, missed this file in the last commit. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@187303 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	fe11c67980	Merged revisions 186321 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r186321 | file | 2009-04-03 12:52:50 -0300 (Fri, 03 Apr 2009) | 12 lines Merged revisions 186320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5 lines Fix a problem with the crypto variable definitions not actually being defined properly. (closes issue #14804) Reported by: jvandal ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@186324 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	05fd775230	Merged revisions 185912 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r185912 | tilghman | 2009-04-01 15:13:28 -0500 (Wed, 01 Apr 2009) | 4 lines Merge changes from str_substitution that are unrelated to that branch. Included is a small bugfix to an ast_str helper, but most of these changes are simply doxygen fixes. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@185947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	2616defc54	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/1.6.2@184779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago