asterisk

Commit Graph

Author	SHA1	Message	Date
BJ Weschke	9aefadd7c1	Do NOT attempt to do anything with the ast_config struct when it's been returned as INVALID by the config file interpreter. (closes issue #13741) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151246 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	1ddc834b39	cleaup of the TCP/TLS socket API: 1) rename 'struct server_args' to 'struct ast_tcptls_session_args', to follow coding guidelines 2) make ast_make_file_from_fd() static and rename it to something that indicates what it really is for (again coding guidelines) 3) rename address variables inside 'struct ast_tcptls_session_args' to be more descriptive (dare i say it... coding guidelines) 4) change ast_tcptls_client_start() to use the new 'remote_address' field of the session args for the destination of the connection, and use the 'local_address' field to bind() the socket to the proper source address, if one is supplied 5) in chan_sip, ensure that we pass in the PP address we are bound to when creating outbound (client) connections, so that our connections will appear from the correct address git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
BJ Weschke	09e9b5d208	Using the GetVar handler in AMI is potentially dangerous (insta-crash [tm]) when you use a dialplan function that requires a channel and then you don't provide one or provide an invalid one in the Channel: parameter. We'll handle this situation exactly the same way it was handled in pbx.c back on r61766. We'll create a bogus channel for the function call and destroy it when we're done. If we have trouble allocating the bogus channel then we're not going to try executing the function call at all and run the risk of crashing. (closes issue #13715) reported by: makoto patch by: bweschke git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150817 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	805556773f	Fix CLI command 'channel request hangup' Prodded on IRC by Russell and fixed by eliel (closes issue #13730) Reported by: eliel Patches: main_cli.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	4ad187cba4	Merged revisions 150304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r150304 \| mmichelson \| 2008-10-16 18:40:54 -0500 (Thu, 16 Oct 2008) \| 6 lines Reverting changes from commits 150298 and 150301 since I was mistakenly under the assumption that dialplan functions always required that a channel be present. I need to go home earlier, I think :) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150305 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	8a1d9d1678	Merged revisions 150298,150301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r150298 \| mmichelson \| 2008-10-16 18:34:37 -0500 (Thu, 16 Oct 2008) \| 10 lines Don't try to call a dialplan function's read callback from the manager's GetVar handler if an invalid channel has been specified. Several dialplan functions, including CHANNEL and SIP_HEADER, do not check for NULL-ness of the channel being passed in. (closes issue #13715) Reported by: makoto ........ r150301 \| mmichelson \| 2008-10-16 18:35:07 -0500 (Thu, 16 Oct 2008) \| 3 lines And don't forget to return on the error condition ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	29a8fe20c8	Merged revisions 149204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149204 \| mmichelson \| 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) \| 12 lines Add a tolerance period for sync-triggered audiohooks so that if packetization of audio is close (but not equal) we don't end up flushing the audiohooks over small inconsistencies in synchronization. Related to issue #13005, and solves the issue for most people who were experiencing the problem. However, a small number of people are still experiencing the problem on long calls, so I am not closing the issue yet ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	d5837ba8c2	Add additional memory debugging to several core APIs, and fix several memory leaks found with these changes. (Closes issue #13505, closes issue #13543) Reported by: mav3rick, triccyx Patches: 20081001__bug13505.diff.txt uploaded by Corydon76 (license 14) Tested by: mav3rick, triccyx git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	b17413c992	Merged revisions 148611 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148611 \| kpfleming \| 2008-10-14 02:54:41 -0500 (Tue, 14 Oct 2008) \| 3 lines it would be nice if this message printing code had actually been tested before it was committed... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	c1351ad237	Merge realtime_update2 branch, which adds a new realtime API call named 'update2', which permits updates which match across multiple columns, instead of requiring all tables to have a single unique identifier. All of the other API calls with the exception of 'update' already had the ability to match on multiple fields, so it was a missing and very desireable feature that an API call implementing an update should have this, too. This does not change any outward performance of Asterisk, but it should make life easier for application developers who use the RealTime framework. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	db7299f4bc	Hmmm. Nobody (but me) is interested in seeing the trie info when they do 'dialplan show ...' (even with debug set to non-zero); so I set up a 'dialplan debug [context]' cli command instead, to explicitly show just the trie info. I even added an extension_exists() call to make sure the trie info is built. I moved the explanatory header to above the extension loop to ensure it only prints once. And it will do this now, whether debug is set or not. I removed the trie printing from the 'dialplan show' command entirely. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	32d93bbc0e	Highlightning even more bugs in the current tcp/tls implementation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	1dedb785ab	Don't include logger.h in asterisk.h by default as it is causing problems building app_voicemail. Instead, include it where it is needed. This turned out to be a relatively minor issue because other headers include logger.h as well. Need to test -addons before merging this back to 1.6.0. (closes issue #13605) Reported by: tomo1657 Patches: 13605_seanbright.diff uploaded by seanbright (license 71) Tested by: mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	9851feb8fb	The priority was unnecessary for the manager atxfer, so it has been removed. Furthermore, now we actually use the Context argument passed to set the transfer context and don't error out if no context is specified. This addresses the actual problems outlined in issue 12158. Regarding the other points brought up, regarding the inability to not transfer to extensions which cannot be represented by DTMF, it is not enough of a constraint that it is worth attempting to rework the feature. (closes issue #12158) Reported by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	6d70f45506	Merged revisions 146026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r146026 \| murf \| 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) \| 18 lines (closes issue #13579) Reported by: dwagner (closes issue #13584) Reported by: dwagner Tested by: murf, putnopvut The thought occurred to me that the res= from the extension spawn was ending up being returned from the bridge. "Thou shalt not poison the return value". Made the change and it appears to allow blind xfers to work as normal. If I'm wrong, reopen the bugs. But it looks good to me! Many thanks to putnopvut for helping me reproduce this! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	8b14e5f493	Reverting format addition for now git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f5d5eb5e19	Fudges for wav16, just like wav49 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	c897b4e630	(closes issue #13139 ) Reported by: krisk84 Tested by: krisk84 This change prevents a call that is placed in the parkinglot to be picked up before the PBX is finished. If another extension dials the parking extension before the PBX thread has completed at minimum warnings will occur about the PBX not properly being terminated. At worst, a crash could occur. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	e235a07376	(closes issue #13557 ) Reported by: nickpeirson Patches: pbx.c.patch uploaded by nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) Tested by: nickpeirson, murf 1. replaced all refs to bzero and bcopy to memset and memmove instead. 2. added a note to the CODING-GUIDELINES 3. add two macros to asterisk.h to prevent bzero, bcopy from creeping back into the source 4. removed bzero from configure, configure.ac, autoconfig.h.in git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	9335b3ad34	Allow people to select the old console behavior of white text on a black background, by using the startup flag '-B'. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	0dc7ac9a6c	Explicitly setting these fields to NULL was done because I wasn't sure if they would be NULL otherwise. Since they will be set automatically, removing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	2ec290b09d	Similar to r143204, masquerade the channel in the case of Park being called from AGI. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146923 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	50923eab5d	This commit squashes together three commits because the wrong approach was originally used. (One of the commits was only one line.) 1) r143204: The main change here was to masquerade the channel if the channel that was to be parked was running a PBX on it. The PBX thread can then maintain full control of the channel (the zombie) as it expects to while allowing the parking thread full control of the real (parked) channel. 2) r143270: Changed park_call_full to hold the parkinglot lock a little longer, which protects the parkeduser struct from being freed out from underneath. Made sure that the parking extension is added to the parking context while holding the lock thereby ensuring that there are no spurious warnings from removal attempts when a hangup occurs while the parking lot is being announced. 3) r143475: (the one liner) compare peer and chan instead of looking at the parked user (pu), which could have possibly already have been freed by the parking thread git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	abc88c1d61	fix some comment placement git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146877 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	7d8d1f50bb	Explicitly set args in park_call_exec NULL so in the case of no options being passed in, there is no garbage attempted to be used. Also, do not set args to unknown value again if there are no options passed in. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146875 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	6855b63d44	Fix a bug with the last item in CLI history getting duplicated when read from the .asterisk_history file (and subsequently being duplicated when written). We weren't checking the result of fgets() which meant that we read the same line twice before feof() actually returned non- zero. Also, stop writing out an extra blank line between each item in the history file, fix a minor off-by-one error, and use symbolic constants rather than a hardcoded integer. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	4b22fb221c	remove superfluous reference counting operations in manage_parkinglot since ao2_interator_next increments the ref count automatically git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146242 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	b29fb615ae	Resolve a subtle bug where we would never successfully be able to get the first item in the CLI entry list. This was preventing '!' from showing up in either 'help' or in tab completion. (closes issue #13578) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	84b0093bef	The dialing API should inherit datastores as well as variables git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	529874de7b	Add schedule extensions to app_meetme. In addition, the reporter found a problem within strptime(3), which we are correcting here with ast_strptime(). (closes issue #11040) Reported by: DEA Patches: 20080910__bug11040.diff.txt uploaded by Corydon76 (license 14) Tested by: DEA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145649 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	7eae109418	Okay, this should really do it now. While I did manage to fix blind transfers with my last commit here, I also caused an unwanted side-effect. That is, only the first priority of the 'h' extension would be executed when a blind transfer occurred instead of all priorities. Essentially, my last commit corrected the return value of ast_bridge_call. However, the implementation still was not 100% correct. Now it is. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	e6799e1c99	if (!(x) == 0) is the same as if (x). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	a23b8c9158	The logic surrounding the return value of ast_spawn_extension within ast_bridge_call was reversed. This problem was observed when a blind transfer placed from the callee channel of a test call failed. While the problem I am solving here is exactly the same as what was reported in issue #13584, the difference is that this fix I am applying is trunk-only. Issue #13584 was reported against the 1.4 branch, and my tests of 1.4's blind transfers appear to work fine. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	90751b16ca	Merge the cli_cleanup branch. This work is done by lmadsen, junky and mvanbaak during AstriDevCon. This is the second audit the CLI got, and this time lmadsen made sure he had _ALL_ modules loaded that have CLI commands in them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	629861a705	Merged revisions 144924-144925 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r144924 \| kpfleming \| 2008-09-27 10:00:48 -0500 (Sat, 27 Sep 2008) \| 6 lines improve header inclusion process in a few small ways: - it is no longer necessary to forcibly include asterisk/autoconfig.h; every module already includes asterisk.h as its first header (even before system headers), which serves the same purpose - astmm.h is now included by asterisk.h when needed, instead of being forced by the Makefile; this means external modules will build properly against installed headers with MALLOC_DEBUG enabled - simplify the usage of some of these headers in the AEL-related stuff in the utils directory ........ r144925 \| kpfleming \| 2008-09-27 10:13:30 -0500 (Sat, 27 Sep 2008) \| 2 lines fix some minor issues with rev 144924 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	579177ae80	Merged revisions 144677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r144677 \| murf \| 2008-09-26 11:47:13 -0600 (Fri, 26 Sep 2008) \| 12 lines (closes issue #13563) Reported by: mnicholson Patches: found1.diff uploaded by mnicholson (license 96) This patch was mainly meant to apply to trunk and 1.6.x, but I'm applying it to 1.4 also, which should be a perfectly harmless fix to the vast majority of users who are not using external switches, but the few who might be affected will not have to go to the pain of filing a bug report. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	e74584ca3c	(closes issue #13557 ) Reported by: nickpeirson The user attached a patch, but the license is not yet recorded. I took the liberty of finding and replacing ALL index() calls with strchr() calls, and that involves more than just main/pbx.c; chan_oss, app_playback, func_cut also had calls to index(), and I changed them out. 1.4 had no references to index() at all. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	8343faed0e	Merged revisions 144066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r144066 \| murf \| 2008-09-23 10:41:49 -0600 (Tue, 23 Sep 2008) \| 29 lines (closes issue #13489) Reported by: DougUDI Tested by: murf (closes issue #13490) Reported by: seanbright Tested by: murf (closes issue #13467) Reported by: edantie Tested by: murf, edantie, DougUDI This crash happens because we are unsafely handling old pointers. The channel whose cdr is being handled, has been hung up and destroyed already. I reorganized the code a bit, and tried not to lose the fork-cdr-chain concepts of the previous code. I now verify that the 'previous' channel (the channel we had when the bridge was started), still exists, by looking it up by name in the channel list. I also do not try to reset the CDR's of channels involved in bridges. Testing shows it solves the crash problem, and should not negatively impact previous fixes involving CDR's generated during/after blind transfers. (The reason we need to reset the CDR's on the "beginning" channels in the first place). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	48ffe6e772	If attempting to free a NULL pointer when MALLOC_DEBUG is set, don't bother searching for a region to free, just immediately exit. This has the dual benefit of suppressing a warning message about freeing memory at (nil) and of optimizing the free() replacement by not having to do any futile searching for the proper region to free. (closes issue #13498) Reported by: pj Patches: 13498.patch uploaded by putnopvut (license 60) Tested by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@143400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	224980b07b	Merged revisions 143337 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r143337 \| mmichelson \| 2008-09-17 13:24:15 -0500 (Wed, 17 Sep 2008) \| 6 lines Allow for "G.729" if offered in an SDP even though it is not RFC 3551 compliant. Some Cisco switches will send this in an SDP, and it doesn't hurt to be able to accept this. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@143340 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	08af5bb312	Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiating when a file is invalid from when a file is missing. This is most important when we have two configuration files. Consider the following example: Old system: sip.conf users.conf Old result New result ======== ========== ========== ========== Missing Missing SIP doesn't load SIP doesn't load Missing OK SIP doesn't load SIP doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid SIP loads incompletely SIP doesn't load Invalid Missing SIP doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So in the case when users.conf doesn't load because there's a typo that disrupts the syntax, we may only partially load users, instead of failing with an error, which may cause some calls not to get processed. Worse yet, the old system would do this with no indication that anything was even wrong. (closes issue #10690) Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	bbf8c3d7cc	When checking for an encoded character, make sure the string isn't blank, first. (Closes issue #13470) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142748 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	8c53dd7f5e	Merged revisions 142740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142740 \| tilghman \| 2008-09-12 11:27:32 -0500 (Fri, 12 Sep 2008) \| 4 lines Don't return a free'd pointer, when a file cannot be opened. (closes issue #13462) Reported by: wackysalut ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142741 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	67f7ac0499	Merged revisions 142675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142675 \| murf \| 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) \| 29 lines Tested by: sergee, murf, chris-mac, andrew, KNK This is a "second attempt" to restore the previous "endbeforeh" behavior in 1.4 and up. In order to capture information concerning all the legs of transfers in all their infinite combinations, I was forced to this particular solution by a chain of logical necessities, the first being that I was not allowed to rewrite the CDR mechanism from the ground up! This change basically leaves the original machinery alone, which allows IVR and local channel type situations to generate CDR's as normal, but a channel flag can be set to suppress the normal running of the h exten. That flag would be set by the code that runs the h exten from the ast_bridge_call routine, to prevent the h exten from being run twice. Also, a flag in the ast_bridge_config struct passed into ast_bridge_call can be used to suppress the running of the h exten in that routine. This would happen, for instance, if you use the 'g' option in the Dial app. Running this routine 'early' allows not only the CDR() func to be used in the h extension for reading CDR variables, but also allows them to be modified before the CDR is posted to the backends. While I dearly hope that this patch overcomes all problems, and introduces no new problems, reality suggests that surely someone will have problems. In this case, please re-open 13251 (or 13289), and we'll see if we can't fix any remaining issues. ** trunk note: some code to suppress the h exten being run from app_queue was added; for the 'continue' option available only in trunk/1.6.x. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142676 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	c4fc8fe4be	Merged revisions 142575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142575 \| murf \| 2008-09-11 16:55:49 -0600 (Thu, 11 Sep 2008) \| 20 lines (closes issue #13364) Reported by: mdu113 Well, fundamentally, the problems revealed in 13364 are because of the ForkCDR call that is done before the dial. When the bridge is in place, it's dealing with the first (and wrong) cdr in the list. So, I wrote a little func to zip down to the first non-locked cdr in the chain, and thru-out the ast_bridge_call, these results are used instead of raw chan->cdr and peer->cdr pointers. This shouldn't affect anyone who isn't forking cdrs before a dial, and should correct the cdr's of those that do. So, this change ends up correcting the dstchannel and userfield; the disposition was fixed by a previous patch, it was OK coming into this problem. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	4fc65a69a2	Merged revisions 142474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142474 \| murf \| 2008-09-10 15:58:17 -0600 (Wed, 10 Sep 2008) \| 30 lines (closes issue #12318) Reported by: krtorio I made a small change to the code that handles local channel situations. In that code, I copy the answer time from the peer cdr, to the bridge_cdr, but I wasn't also copying the disposition from the peer cdr. So, Now I copy the disposition, and I've tested against these cases: 1. phone 1 never answers the phone; no cdr is generated at all. this should show up as a manager command failure or something. 2. phone 2 never answers. CDR is generated, says NO ANSWER 3. phone 2 is busy. CDR is generated, says BUSY 4. phone 2 answers: CDR is generated, times are correct; disposition is ANSWERED, which is correct. The start time is the time that the manager dialed the first phone. The answer time is the time the second phone picks up. I purposely left the cid and src fields blank; since this call really originates from the manager, there is no 'easy' data to put in these fields. If you feel strongly that these fields should be filled in, re-open this bug and I'll dig further. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142475 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	bdfd731b8b	Merged revisions 142354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142354 \| russell \| 2008-09-10 11:39:53 -0500 (Wed, 10 Sep 2008) \| 7 lines It is a normal situation that a task gets put in the scheduler that should run as soon as possible. Accept "0" as an acceptable time to run, and also treat negative as "run now", and don't print a debug message about it. (inspired by a message asking about the "request to schedule in the past" debug message on the -dev list) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	75e95cb2b3	Cleaned up comment git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142181 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	4f077691bc	Merged revisions 142063 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142063 \| russell \| 2008-09-09 10:40:24 -0500 (Tue, 09 Sep 2008) \| 5 lines Ensure that the stored CDR reference is still valid after the bridge before poking at it. Also, keep the channel locked while messing with this CDR. (fixes crashes reported in issue #13409) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142064 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	fbe13cfb86	Modify ast_answer() to not hold the channel lock while calling ast_safe_sleep() or when calling ast_waitfor(). These are inappropriate times to hold the channel lock. This is what has caused "could not get the channel lock" messages from chan_sip and has likely caused a negative impact on performance results of SIP in Asterisk 1.6. Thanks to file for pointing out this section of code. (closes issue #13287) (closes issue #13115) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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