asterisk

Commit Graph

Author	SHA1	Message	Date
Mark Michelson	c1e2636be7	Add missing comment that quotes RFC 3891 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	85a5f68fe1	Merged revisions 181029,181031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181029 \| mmichelson \| 2009-03-10 19:30:26 -0500 (Tue, 10 Mar 2009) \| 9 lines Fix incorrect tag checking on transfers when pedantic=yes is enabled. (closes issue #14611) Reported by: klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ r181031 \| mmichelson \| 2009-03-10 19:32:40 -0500 (Tue, 10 Mar 2009) \| 3 lines Remove unused variables. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	bfc0d3b795	Add MALLOC_DEBUG to various utility APIs, so that memory leaks can be tracked back to their source. (related to issue #14636) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ac7e490b94	Spacing changes only git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	1322112da0	Merged revisions 180941 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180941 \| qwell \| 2009-03-10 17:02:18 -0500 (Tue, 10 Mar 2009) \| 1 line Make things happier when using autoconf 2.62+ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180944 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	4b9a0c8aed	Add some notes on getting in contact with the dev community git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0576d57d49	Remove difficulty and language specifiers git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	e3a339512e	Expand upon documentation of manager event project git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180935 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	eddf496f3a	list the move of the astvarrundir from /var/run to /var/run/asterisk (actually its $(localstatedir)/run/asterisk Makes setups with asterisk as non-root easier to manage because you can setup permissions on this dir instead of touching a file and setting permissions on that. Files that come to mind are asterisk.pid and asterisk.ctl socket. Prodded by and ok @russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180898 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	093b469ef5	add more projects git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180862 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	04fbb5aaa6	add more project ideas git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180859 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	951cbf11d4	Reset the thread local string buffer when handling the UserEvent action. (closes issue #14593) Reported by: JimDickenson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180800 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	91fef42422	Add current mentors list, and first pass on a project list broken out of "PineMango" I will work on adding projects that have been sent to be via email tomorrow. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180750 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	bf0bb7b385	Add Doxygen documentation for API changes from 1.6.0 to 1.6.1 Copied from my review board description: This is a continuation of the API changes documentation started for describing changes between releases. Most of the API changes were pretty simple needing only to be brought to attention via the new "Asterisk API Changes" list. However, if you see anything that needs further explanation feel free to supplement what is there. The current method of documenting is to add (in the header file): \version <ver number> <description of changes> and then to add the function to the change list in doxyref.h on the AstAPIChanges page. I also made sure all the functions that were newly added were tagged with \since 1.6.1. I think this is a good habit to start both for the historical aspect as well as for the future ability to easily add a "New Asterisk API" page. Review: http://reviewboard.digium.com/r/190/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	dea550a292	Add skeleton for GSoC ideas list git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	34ae4d2825	Make some minor updates to the doxygen configuration - add bridges directory to be processed - add some res/ subdirs - alphabetize subdirs - use consistent indentation git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	09df92d485	Merged revisions 180567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180567 \| mmichelson \| 2009-03-06 12:23:09 -0600 (Fri, 06 Mar 2009) \| 2 lines Make compilation succeed in dev-mode when IMAP storage is enabled. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	02de67c232	Merged revisions 180532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180532 \| dvossel \| 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009) \| 9 lines Fix handling of backreferences for ENUM lookups enum.c did not handle regex backtraces correctly. The '\1' in the regex is a backreference that requires a pattern match to be inserted. The way the code used to work is that it would find the backreference and insert the entire input string minus the '+'. This is incorrect. The regexec() function takes in a variable called pmatch which is an array of structs containing the start and end indexes for each backreference substring. The original code actually passed the pmatch array pointer into regexec but never did anything with it. Now when a backtrace is found, the backtrace number is looked up in the pmatch array and the correct substring is inserted. (closes issue #14576) Reported by: chris-mac Review: http://reviewboard.digium.com/r/187/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	96405af1a8	Merged revisions 180464 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180464 \| mmichelson \| 2009-03-05 17:26:11 -0600 (Thu, 05 Mar 2009) \| 16 lines [IMAP] Fix message retrieval issues when identical mailbox names were defined in separate contexts. There was a fix put in a while back so that an X-Asterisk-VM-Context message header was added to stored IMAP voicemails. This would allow for us to differentiate if the same mailbox name was used in multiple contexts. The problem still left was that not all places where messages were retrieved actually attempted to use this header for information when retrieving messages. This commit fixes that so that MWI and message retrieval from VoiceMailMain work as expected. (closes issue #13853) Reported by: vicks1 Patches: 13853_v2.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	e69803a2be	Merged revisions 180380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180380 \| mmichelson \| 2009-03-05 12:58:48 -0600 (Thu, 05 Mar 2009) \| 25 lines Fix broken mailbox parsing when searchcontexts option is enabled. When using the searchcontexts option in voicemail.conf, the code made the assumption that all mailbox names defined were unique across all contexts. However, the code did nothing to actually enforce this assumption, nor did it do anything to alert a user that he may have created an ambiguity in his voicemail.conf file by defining the same mailbox name in multiple contexts. With this change, we now will issue a nice long warning if searchcontexts is on and we encounter the same mailbox name in multiple contexts and ignore any duplicates after the first box. Whether searchcontexts is enabled or not, if we come across a duplicate mailbox in the same context, then we will issue a warning and ignore the duplicated mailbox. I have also added a small note to voicemail.conf.sample in the explanation for searchcontexts explaining that you cannot define the same mailbox in multiple contexts if you have enabled the option. (closes issue #14599) Reported by: lmadsen Patches: 14599.patch uploaded by mmichelson (license 60) (with slight modification) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	9348bfd926	Make sure we terminate the first s\| command so we can actually produce correct files. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	2f24689b49	Merged revisions 180372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180372 \| kpfleming \| 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) \| 9 lines Fix problems when RTP packet frame size is changed During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180373 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	4c9ab0df8c	Merge phase 1 support for the new bridging architecture. This commit brings in the bridging core, bridging technologies, and the ConfBridge application. For usage information on the ConfBridge application please see the output of "core show application ConfBridge" from the CLI. For API documentation please see the doxygen page describing the architecture and the documentation for each API call. Review: http://reviewboard.digium.com/r/93/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	dd1a5f1969	Also highlight the preamble and postamble git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180334 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	2c54fc25cd	Add syntax coloring files for Vim, including a new one for AEL git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	6c9f6d33c7	Resolve object matching issues related to the removal of the sip_user object. Previously, chan_sip had both sip_peer and sip_user objects in memory. A patch went in to remove sip_user to simplify the code, since everything could be done with just sip_peer. This patch resolves some regressions found that were introduced by those changes. This code comes from svn/asterisk/team/group/sip-object-matching/. Here is a list of the changes that have been made: 1) When doing a match by name with the find_peer() function, make it much easier to specify which objects should be matched by having a parameter that specifies exactly which object types should be considered. Also, update find_by_name() to handle this parameter. Finally, update all code to use the new option values. 2) When looking up an object for an outbound request by name, consider peers only. (create_addr()) 3) Only match peers on an incoming registration request. 4) When doing authentication (except for SUBSCRIBE), look up users by name, instead of all objects by name. 5) When doing authentication (except for SUBSCRIBE), after looking for a user by name, look for a peer by IP address, instead of all objects by IP address. 6) When handling the SIP qualify CLI command or manager action, look for a peer by name, instead of any object by name. 7) When handling the SIP unregister CLI command, look for a peer by name, instead of any object by name. 9) In sip_do_debug_peer(), search for a peer by name, instead of any object by name. 9) When handling the SIPPEER() dialplan function, search for a peer by name, instead of any object by name. 10) In the following session timer related functions, st_get_se(), st_get_refresher(), and st_get_mode(), when looking for an object for a given sip_pvt using pvt->peername, look for a peer by name, instead of any object by name. 11) Fix build_peer() to properly handle the case where separate type=peer and type=user entries were specified in sip.conf. (closes issue #14505) Reported by: lmadsen Review: http://reviewboard.digium.com/r/172/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	eb5bb03b82	Spacing changes only git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	a66032a14a	Merged revisions 180194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180194 \| file \| 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) \| 4 lines Look for the number in a callerid string starting from the end. This way a value using <> can exist in the name portion. (issue #AST-194) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180195 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	3a14487abf	Allow for "magic" pickups to work when we wish to ignore the context When the subscription context for a call pickup subscription differs from the context of the call pickup target, there's not an easy way to divine what context should be used for the pickup. The way to work around this is to use PICKUPMARK as the context for the pickup. This has been documented in the sip.conf.sample file (ABE-1708) closes issue #14567 submitted by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	15090ba1df	Remove duplicate 'k' and 'K' Dial options. (closes issue #14601) Reported by: alecdavis Patches: app_dial.optionk.diff.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	ceea9b1dce	My bad! left check_expr2 in the ALL_UTILS list by mistake. Already done to 1.6.x git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	979eb709ae	app_read does not break from prompt loop with user terminated empty string In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata(). (closes issue #14279) Reported by: Marquis Patches: fix_app_read.patch uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	0b01444ab7	Blocked revisions 180010 via svnmerge ........ r180010 \| qwell \| 2009-03-03 17:01:06 -0600 (Tue, 03 Mar 2009) \| 1 line Make sure we still support zapchan in users.conf, in addition to dahdichan. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	8970f8caaa	Merged revisions 180006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180006 \| mmichelson \| 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) \| 17 lines Clarify some documentation of queues.conf.sample It had always been possible to explicitly specify a "blank" value for a sound file in queues.conf and have no sound played back. The problem with this is that it would result in some ugly CLI warnings from file.c. This commit introduces a check when playing a file in app_queue to see if the name of the file is zero-length and return early if that is the case. Also, the ability to specify the blank sound files in queues.conf is now mentioned more clearly in queues.conf.sample (closes issue #14227) Reported by: caspy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	f47b03877b	Merged revisions 179807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some work to do to port these changes to trunk; the check_expr stuff hasn't been updated here for quite some time, it appears. I added some more tests to the check_expr2 suite. I had to play around with the makefile a bit, etc. I added STANDALONE2 #ifdefs to ast_expr2.y so as not to conflict structure with aelparse. ........ r179807 \| murf \| 2009-03-03 11:11:34 -0700 (Tue, 03 Mar 2009) \| 19 lines These changes allow AEL to better check ${} constructs within $[...], that are concatenated with text. I modified and added rules in ast_expr2.fl to better handle the concatenations. I added some default routines to ast_expr2.y so the standalone would compile. It also looks like I haven't run this thru bison since 2.1, so it's good to get this updated. The Makefile has comments added now for check_expr2 and check_expr to explain what they are for, and how to run them. The testexpr2s stuff has been removed, in favor of check_expr2. expr2.testinput has been updated to include the two expressions that inspired these changes (from mcnobody on #asterisk this morning) The regression has been run and all looks well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179973 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	ae786501f1	app_meetme not setting filename and fileformat correctly for realtime When app_meetme finds a realtime conference, it doesn't get the filename and fileformat correctly when 'r' is set. Now app_meetme first checks to see if fileformat and filename are declared in the db, if they're not it checks the .conf file, if its not declared there either it then uses defaults. (closes issue #14545) Reported by: dalbaech Patches: app_meetme-realtime5.patch uploaded by dvossel (license 671) Realtime_Conference_Record_workaround.txt uploaded by dalbaech (license 705) Tested by: dvossel, dalbaech Review: http://reviewboard.digium.com/r/180/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179972 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	d8d5e38f65	Add documentation for timing modules used in Asterisk This document specifies the timing modules available in Asterisk beginning with Asterisk 1.6.1. The document goes into detail about the differences between each and gives a general overview of what timing is used for in Asterisk. There is also a section which can be used to help customize your setup or to troubleshoot timing issues you may have. I also added messages to the DAHDI timing test used in res_timing_dahdi.c that points to this new documentation if people experience problems. Big thanks to all who contributed comments on this. (closes issue #14490) Reported by: mmichelson Patches: timing.txt uploaded by mmichelson (license 60) Review: http://reviewboard.digium.com/r/164/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179937 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Brian Degenhardt	a7092f0acc	fix a leaked channel lock (and future deadlock) when we try to pick up our own channel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179903 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	bcf5ecde90	Merged revisions 179840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179840 \| file \| 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) \| 9 lines Do not assume that the bridge_cdr is still attached to the channel when the 'h' exten is finished executing. It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to. We can not safely modify it afterwards because of this, so don't even try. (closes issue #14564) Reported by: meric ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	7824d71998	Convert pbx_spool to use string fields instead of statically-sized buffers. In tests run after making this conversion, I noticed an approximate 85% reduction in memory usage for call file processing. Review: http://reviewboard.digium.com/r/168/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179745 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	cfa0d9c0ce	Merged revisions 179741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179741 \| russell \| 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) \| 6 lines Ensure chan->fdno always gets reset to -1 after handling a channel fd event. Since setting fdno to -1 had to be moved, a couple of other code paths that do process an fd event return early and do not pass through the code path where it was moved to. So, set it to -1 in a few other places, too. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179742 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	f000d5bb0f	Please prefix default values with DEFAULT git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	a65727949c	Merged revisions 179671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179671 \| file \| 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) \| 3 lines Move where fdno is set to the default value to after the read callback of the channel driver is called. We have to do this as the underlying channel driver may need the fdno value to determine what to read. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	d9b034a430	Merged revisions 179608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179608 \| russell \| 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) \| 9 lines Make it easier to detect an improper call to ast_read(). When you call ast_waitfor() on a channel, the index into the channel fds array that holds the file descriptor that poll() determines has input available is stored in fdno. This patch clears out this value after a call to ast_read() and also reports errors if ast_read() is called without an fdno set. From a discussion on the asterisk-dev list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179609 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	aa81288bab	Merged revisions 179536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179536 \| jpeeler \| 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) \| 15 lines Fix bridging regression from commit 176701 This fixes a bad regression where the bridge would exit after an attended transfer was made. The problem was due to nexteventts getting set after the masquerade which caused the bridge to return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: tim_ringenbach ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8f6a933e30	Merged revisions 179532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179532 \| russell \| 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) \| 40 lines Move ast_waitfor() down to avoid the results of the API call becoming stale. This call to ast_waitfor() was being done way too soon in this section of code. Specifically, there was code in between the call to waitfor and the code that uses the result that puts the channel in autoservice. By putting the channel in autoservice, the previous results of ast_waitfor() become meaningless, as the autoservice thread will do it's own ast_waitfor() and ast_read() on the channel. So, when we came back out of autoservice and eventually hit the block of code that calls ast_read() on the channel, there may not actually be any input on the channel available. Even though the previous call to ast_waitfor() in app_meetme said there was input, the autoservice thread has since serviced the channel for some period of time. This bug manifested itself while dvossel was doing some testing of MeetMe in Asterisk trunk. He was using the timerfd timing module. When the code hit ast_read() erroneously, it determined that it must have been called because of input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was the cause of the last legitimate call to ast_read() done by autoservice. In this test, an IAX2 channel was calling into the MeetMe conference. It was _much_ more likely to be seen with an IAX2 channel because of the way audio is handled. Every audio frame that comes in results in a call to ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify the channel thread that a frame is waiting to be handled. So, the chances of ast_waitfor() indicating that a channel needs servicing due to a timer event on an IAX2 event is very high. Finally, it is interesting to note that if a different timing interface was being used, this bug would probably not be noticed. When ast_read() is called and erroneously thinks that there is a timer event to handle, it calls the ast_timer_ack() function. The pthread and dahdi timing modules handle the ack() function being called when there is no event by simply ignoring it. In the case of the timerfd module, it results in a read() on the timer fd that will block forever, as there is no data to read. This caused Asterisk to lock up very quickly. Thanks to dvossel and mmichelson for the fun debugging session. :-) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	3252cd2e5b	Merged revisions 179468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179468 \| tilghman \| 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) \| 10 lines When ending a recording with silence detection, remember to reduce the duration. The end of the recording is correspondingly trimmed, but the duration was not trimmed by the number of seconds trimmed, so the saved duration was necessarily longer than the actual soundfile duration. (closes issue #14406) Reported by: sasargen Patches: 20090226__bug14406.diff.txt uploaded by tilghman (license 14) Tested by: sasargen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	d6652e9760	Fix a reference leak in timerfd_set_rate(). (found during a debugging session with dvossel and mmichelson.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0c0479602e	Merged revisions 179461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179461 \| russell \| 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) \| 8 lines Ensure that only one thread is calling ast_settimeout() on a channel at a time. For example, with an IAX2 channel, you can have both the channel thread and the chan_iax2 processing threads calling this function, and doing so twice at the same time is a bad thing. (Found in a debugging session with dvossel and mmichelson) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	9bb9c64521	Merged revisions 179395 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179395 \| qwell \| 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) \| 1 line Remove several silly warnings in editline. One about a broken preprocessor directive, and another about strlcpy/strlcat. (closes issue #14264) Reported by: dimas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179396 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

... 3 4 5 6 7 ...

17773 Commits (f6de93c5dfeb7d54ded98a3f364dc54703ccc06c) All Branches Search

17773 Commits (f6de93c5dfeb7d54ded98a3f364dc54703ccc06c)

All Branches