2017 Commits (1.6.2)

Author	SHA1	Message	Date
Richard Mudgett	9172658d42	Merged revisions 318734 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r318734 | rmudgett | 2011-05-12 20:09:40 -0500 (Thu, 12 May 2011) | 43 lines Merged revisions 318671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 * The applicable fixes for v1.4 are the SIP deadlock and the in progress masquerade check for multiple parties trying to pickup the same call. issue18654_v1.4.patch uploaded by rmudgett (license 664) * Backported to v1.6.2. issue18654_v1.6.2.patch uploaded by rmudgett (license 664) ........ r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines Fix directed group pickup feature code *8 with pickupsounds enabled Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues. 1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked. 2). dialplan applications for directed_pickups shouldn't beep. 3). feature code for directed pickup should beep on success/failure if configured. Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite. Moved app_directed:pickup_do() to features:ast_do_pickup(). Functions below, all now use the new ast_do_pickup() app_directed_pickup.c: pickup_by_channel() pickup_by_exten() pickup_by_mark() pickup_by_part() features.c: ast_pickup_call() (closes issue #18654) Reported by: Docent Patches: ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett Review: https://reviewboard.asterisk.org/r/1185/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@318735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	93351f0d0b	Merged revisions 315501 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011) | 14 lines Fix the bounds-checking code. The code that set the bit within the select bitfield was correct, but the bounds-checking code was not. The change to that line uses the new _bitsize macro for clarity. Also, FD_ZERO macro did not zero-out anything but the first word of the bitfield, so this could have caused problems with modules using that macro with the expanded bitfield. (closes issue #18773) Reported by: jamicque Patches: 20110423__issue18773.diff.txt uploaded by tilghman (license 14) Tested by: chris-mac ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@315502 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	1fd86131c5	Merged revisions 313277 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313277 | lmadsen | 2011-04-11 14:30:20 -0500 (Mon, 11 Apr 2011) | 6 lines Fix detection of OpenSSL 1.0 (closes issue #19093) Reported by: tzafrir Patches: detect_openssl_10.diff uploaded by tzafrir (license 46) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@313278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	389b2a6907	Merged revisions 312573 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011) | 38 lines Issues with ISDN calls changing B channels during call negotiations. The handling of the PROCEEDING message was not using the correct call structure if the B channel was changed. (The same for PROGRESS.) The call was also not hungup if the new B channel is not provisioned or is busy. * Made all call connection messages (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS, ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are using the correct structure and B channel. If there is any problem with the operations then the call is now hungup with an appropriate cause code. * Made miscellaneous messages (INFORMATION, FACILITY, NOTIFY) find the correct structure by looking for the call and not using the channel ID. NOTIFY is an exception with versions of libpri before v1.4.11 because a call pointer is not available for Asterisk to use. * Made all hangup messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find the correct structure by looking for the call and not using the channel ID. (closes issue #18313) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2620 (closes issue #18231) Reported by: destiny6628 Tested by: rmudgett JIRA SWP-2924 (closes issue #18488) Reported by: jpokorny JIRA SWP-2929 JIRA AST-437 (The issues fixed here are most likely causing this JIRA issue.) JIRA DAHDI-406 JIRA LIBPRI-33 (Stuck resetting flag likely fixed) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@312574 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	82f042979f	Merged revisions 312285 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011) | 7 lines Found some leaking file descriptors while looking at ast_FD_SETSIZE dead code. (issue #18969) Reported by: oej Patches: 20110315__issue18969__14.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@312287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	85a049ff26	Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround. ... Not surprisingly, the workaround was exactly the same code as was provided by the Flex maintainers, albeit in two different places, in different macros. This should fix the FreeBSD builds, which have an older version of Flex. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@309251 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	d8bc2281bb	A later version of flex already includes the fwrite workaround code, which if used twice causes a compilation error. ... Detect whether Flex will compile without the workaround; if so, suppress our workaround code. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@309033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	29469758df	Merge missing bugfix for issue #11583 ... This is the combination of two commits that made it into 1.4, 1.6.0, 1.6.1, and trunk (and therefor 1.8) but that was missed for 1.6.2. ........ r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines Cleaning up a few things in detect disconnect patch Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect. issue #11583 ........ ........ r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines Allow disconnect feature before a call is bridged feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c. (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) detect_disconnect.diff uploaded by dvossel (license 671) Tested by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@308677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	0f8989a993	Merged revisions 303546 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines Fix channel redirect out of MeetMe() and other issues with channel softhangup. Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped working properly. This issue includes a patch that resolves the issue by removing a call to ast_check_hangup() from app_meetme.c. I left that in my patch, as it doesn't need to be there. However, the rest of the patch fixes this problem with or without the change to app_meetme. The key difference between what happens before and after this patch is the effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(), ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme sees this which causes it to exit as intended. Checking ast_check_hangup() caused app_meetme to exit earlier in the process, and the target of the redirect saw the condition where ast_read() returned NULL. Removing ast_check_hangup() works around the issue in app_meetme, but doesn't solve the issue if another application did the same thing. There are also other edge cases where if an application finishes at the same time that a redirect happens, the target of the redirect will think that the channel hung up. So, I made some changes in pbx.c to resolve it at a deeper level. There are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to abort the hangup process. My patch extends this to remove the END_OF_Q frame from the channel's read queue, making the "abort hangup" more complete. This same technique was used in every place where a softhangup flag was cleared. (closes issue #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: https://reviewboard.asterisk.org/r/1082/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303548 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	5d7f358fc2	Merged revisions 302663 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302663 | tilghman | 2011-01-19 15:20:28 -0600 (Wed, 19 Jan 2011) | 2 lines Add some API documentation ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	fb52725167	Merged revisions 298905 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010) | 6 lines Let Asterisk find better backtrace information with libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will use libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@298957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	fd387d7491	Portability issue on OpenSolaris. ... Also detect the required structure element, because OpenSolaris defines SIOCGIFHWADDR, but without support for IP sockets. (closes issue #18442) Reported by: ranjtech Patches: 20101209__issue18442.diff.txt uploaded by tilghman (license 14) Tested by: ranjtech git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@298050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	1ff4cc9dc7	Merged revisions 296990 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01 Dec 2010) | 5 lines Clarify documentation on how we store codec preference lists. (closes issue #18397) Reported by: birgita ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@296991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	cade893d5a	I love standards. There are so many to choose from. Except when there isn't one. ... Linux and *BSD disagree on the elements within the ucred structure. Detect which one is in use on the system. (closes issue #18384) Reported by: bjm Patches: cred-diffs uploaded by bjm (license 473) 20101127__issue18384__1.6.2.diff.txt uploaded by tilghman (license 14) 20101127__issue18384__1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, bjm git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@296533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	da2b2990bf	Merged revisions 295790 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. To recreate the problem: 1) Party A calls Party B 2) Invoke CLI "channel redirect" command to redirect channel call leg associated with A. 3) All associated channels are hung up. Note that if the CLI command were done on the channel call leg associated with B it works. This regression was a result of the fix for issue #16946 (https://reviewboard.asterisk.org/r/740/). The regression affects all features that use an async goto to execute the dialplan because of an external event: Channel redirect, AMI redirect, SIP REFER, and FAX detection. The struct ast_channel._softhangup code is a mess. The variable is used for several purposes that do not necessarily result in the call being hung up. I have added doxygen comments to describe how the various _softhangup bits are used. I have corrected all the places where the variable was tested in a non-bit oriented manner. The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so the soft hangup requests that do not normally result in a hangup do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171) Reported by: SantaFox (closes issue #18185) Reported by: kwemheuer (closes issue #18211) Reported by: zahir_koradia (closes issue #18230) Reported by: vmarrone (closes issue #18299) Reported by: mbrevda (closes issue #18322) Reported by: nerbos Review: https://reviewboard.asterisk.org/r/1013/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@295843 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	94597cd873	Fix cache of device state changes for multiple servers. ... This patch addresses a regression where device states across multiple servers were not being processing completely correctly. The code works to determine the overall state by looking at the last known state of a device on each server. However, there was a regression due to some invasive rewrites of how the cache works that led to the cache only storing the last device state change for a device, regardless of which server it was on. The code is set up to cache device state change events by ensuring that each event in the cache has a unique device name + entity ID (server ID). The code that was responsible for comparing raw information elements (which EID is) always returned a match due to a memcmp() with a length of 0. There isn't much code to fix the actual bug. This patch also introduces a new CLI command that was very useful for debugging this problem. The command allows you to dump the contents of the event cache. (closes issue #18284) Reported by: klaus3000 Patches: issue18284.rev1.txt uploaded by russell (license 2) Tested by: russell, klaus3000 (closes issue #18280) Reported by: klaus3000 Review: https://reviewboard.asterisk.org/r/1012/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@295710 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	952edec236	Fix compiler warnings when using openssl-dev 1.0.0+ ... Review: https://reviewboard.asterisk.org/r/1016/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@295440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	d96d0ac711	Merged revisions 294384 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010) | 47 lines Fix a deadlock in device state change processing. Copied from some notes from the original author (Russell): Deadlock scenario: Thread 1: device state change thread Holds - rdlock on contexts Holds - hints lock Waiting on channels container lock Thread 2: SIP monitor thread Holds the "iflock" Holds a sip_pvt lock Holds channel container lock Waiting for a channel lock Thread 3: A channel thread (chan_local in this case) Holds 2 channel locks acquired within app_dial Holds a 3rd channel lock it got inside of chan_local Holds a local_pvt lock Waiting on a rdlock of the contexts lock A bunch of other threads waiting on a wrlock of the contexts lock To address this deadlock, some locking order rules must be put in place and enforced. Existing relevant rules: 1) channel lock before a pvt lock 2) contexts lock before hints lock 3) channels container before a channel What's missing is some enforcement of the order when you involve more than any two. To fix this problem, I put in some code that ensures that (at least in the code paths involved in this bug) the locks in (3) come before the locks in (2). To change the operation of thread 1 to comply, I converted the storage of hints to an astobj2 container. This allows processing of hints without holding the hints container lock. So, in the code path that led to thread 1's state, it no longer holds either the contexts or hints lock while it attempts to lock the channels container. (closes issue #18165) Reported by: antonio ABE-2583 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@294639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	b94c33af8c	Fix playback failure when using IAX with the timerfd module. ... To fix this issue the alert pipe will now be used when the timerfd module is in use. There appeared to be a race that was not solved by adding locking in the timerfd module, but needed to be there anyway. The race was between the timer being put in non-continuous mode in ast_read on the channel thread and the IAX frame scheduler queuing a frame which would enable continuous mode before the non-continuous mode event was read. This race for now is simply avoided. (closes issue #18110) Reported by: tpanton Tested by: tpanton I put tested by tpanton because it was tested on his hardware. Thanks for the remote access to debug this issue! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@294277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	77f01ee757	Merged revisions 290750 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) | 9 lines Allow PRI to build properly when using --with-pri. Use the directories found for the parent when using lib dependencies. (closes issue #17314) Reported by: tzafrir Patches: 17314-withdeps.diff uploaded by qwell (license 4) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@290751 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	697b634b47	Merged revisions 289797 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines Change RFC2833 DTMF event duration on end to report actual elapsed time. The scenario here is with a non P2P early media session. The reported time length of DTMF presses are coming up short when sending to the remote side. Currently the event duration is a running total that is incremented when sending continuation packets. These continuation packets are only triggered upon incoming media from the remote side, which means that the running total probably is not going to end up matching the actual length of time Asterisk received DTMF. This patch changes the end event duration to be lengthened if it is detected that the end event is going to come up short. Review: https://reviewboard.asterisk.org/r/957/ ABE-2476 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@289798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	20c729d409	Merged revisions 288636 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23 Sep 2010) | 2 lines Solaris compatibility fixes ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@288637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	a6ecd54058	Merged revisions 286059 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines Inherit CHANNEL() writes to both sides of a Local channel Having Local (/n) channels as queue members and setting the language in the extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2 channel. Hold time report playbacks happen on the Local/...,1 channel and therefor do not play in the specified language. This patch modifies func_channel_write to call the setoption callback and pass the CHANNEL() write info to the callback. chan_local uses this information to look up the other side of the channel and apply the same changes to it. (closes issue #17673) Reported by: Guggemand Review: https://reviewboard.asterisk.org/r/903/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@286115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	6e26576166	Another fix for Mac OS X. ... While trying to fix this the "right" way, I wandered into dependency hell. Two hours later, I backed out, and just removed the offending code. ast_inline_api only goes one level deep and then it breaks. Ouch. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285961 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8d4cda01cb	Merged revisions 285889 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) | 7 lines Fix Mac OS X build. This also fixes a rather grievous calculation error for the offset of ast_fdset, which was masked on Linux and FreeBSD, because these platforms check the first 256 FDs regardless of the bitmask setting (due to backwards compatibility). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285930 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	878abe8339	Follow coding guidelines in moh rescan fix. Also fix the documentation that got me in trouble. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	5e12dc60f2	Failed to rerun bootstrap.sh after last commit ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	59f700a9b2	Merged revisions 284478 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) | 11 lines Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows. This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing a potential crash bug in all supported releases. (closes issue #17678) Reported by: russell Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select Review: https://reviewboard.asterisk.org/r/824/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284593 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	5da0ea7b8e	Merged revisions 280982 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010) | 8 lines Change context lock back to a mutex, because functionality depends upon the lock being recursive. (closes issue #17643) Reported by: zerohalo Patches: 20100726__issue17643.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280983 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3043c8f34e	Merged revisions 279945 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) | 19 lines remove empty audiohook write list on channel If a channel has an audiohook write list created on it, that list stays on the channel until the channel is destroyed. There is no reason to keep that list on the channel if it becomes empty. If it is empty that just means we are doing needless translating for every ast_read and ast_write. This patch removes the audiohook list from the channel once it is detected to be empty on either a read or write. If a audiohook is added back to the channel after this list is destroyed, the list just gets recreated as if it never existed to begin with. (closes issue #17630) Reported by: manvirr Review: https://reviewboard.asterisk.org/r/799/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	19ecaa8b4d	Really fix sounds Makefile (and make it readableish). ... There was a rather large syntax error that should have caused ALL versions of GNU make to fail. I don't know how it worked. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@279657 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	0bf59fb242	Merged revisions 278272 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r278272 | tilghman | 2010-07-20 17:26:23 -0500 (Tue, 20 Jul 2010) | 11 lines Merged revisions 278167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010) | 4 lines Do not queue up DTMF frames while a call is on hold. (Fixes ABE-2110) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@278273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	7e4192b3ae	Merged revisions 277775 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r277775 | tilghman | 2010-07-17 12:42:32 -0500 (Sat, 17 Jul 2010) | 12 lines Merged revisions 277738 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010) | 5 lines Remove uclibc cross-compile triplet, as uclibc has a working fork()... it's only uclinux that does not. (closes issue #17616) Reported by: pprindeville ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@277776 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	5177d85be1	Merged revisions 275022 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r275022 | russell | 2010-07-09 10:35:53 -0500 (Fri, 09 Jul 2010) | 11 lines Merged revisions 275021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010) | 4 lines Document that a leading and trailing slash is expected for test categories. Also, emit a warning if a test is registered without one of these. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@275023 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c25cc0c6cf	Merged revisions 273830 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r273830 | tilghman | 2010-07-02 21:36:31 -0500 (Fri, 02 Jul 2010) | 16 lines Merged revisions 273793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs. (closes issue #17407) Reported by: pdf Patches: 20100527__issue17407.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/751/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@273831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	6ed96a542a	Backport unit test API from trunk. ... Also, update existing test modules that were already in this branch but had been converted to the unit test API in trunk. Review: https://reviewboard.asterisk.org/r/748/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@272531 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	444956b09f	Merged revisions 271690 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r271690 | mnicholson | 2010-06-22 07:58:28 -0500 (Tue, 22 Jun 2010) | 18 lines Merged revisions 271689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun 2010) | 8 lines Modify chan_sip's packet generation api to automatically calculate the Content-Length. This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated. This change was made to ensure that the Content-Length is always correct. (closes issue #17326) Reported by: kenner Tested by: mnicholson, kenner Review: https://reviewboard.asterisk.org/r/693/ ........ This change also adds an ast_str_copy_string() function (similar to ast_copy_string), that copies one ast_str into another, properly handling embedded nulls. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@271691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	bef7474554	Merged revisions 271483 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r271483 | jpeeler | 2010-06-18 16:32:09 -0500 (Fri, 18 Jun 2010) | 18 lines Merged revisions 271399 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010) | 11 lines Fix crash when parsing some heavily nested statements in AEL on reload. Due to the recursion used when compiling AEL in gen_prios, all the stack space was being consumed when parsing some AEL that contained nesting 13 levels deep. Changing a few large buffers to be heap allocated fixed the crash, although I did not test how many more levels can now be safely used. (closes issue #16053) Reported by: diLLec Tested by: jpeeler ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@271484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	2fd6cbc156	Merged revisions 269417 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r269417 | russell | 2010-06-09 16:11:43 -0500 (Wed, 09 Jun 2010) | 6 lines Resolve an invalid memory read on an event. Valgrind pointed out that attempting to get an IE value from an event that has no IEs produces an invalid memory read past the end of the event. Thanks to mmichelson for pointing the problem out to me and then testing the fix. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@269418 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	1a4abf2895	Fix some chan_oss build system problems (these changes were already in trunk). ... This is what caused a bunch of tests to fail on 1.6.2. They expected a console channel driver, but chan_oss was failing to load. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@268815 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f5297544fc	Merged revisions 267862,267877 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r267862 | tilghman | 2010-06-03 21:58:55 -0500 (Thu, 03 Jun 2010) | 5 lines As signed linear audio data is accessed as 16-bit values, certain processors require the values to be aligned in memory. (closes issue #16912) Reported by: michaelevdokimov ........ r267877 | tilghman | 2010-06-03 22:20:47 -0500 (Thu, 03 Jun 2010) | 8 lines As signed linear audio data is accessed as 16-bit values, certain processors require the values to be aligned in memory. (closes issue #16912) Reported by: michaelevdokimov Patches: asterisk.patch uploaded by michaelevdokimov (license 997) Tested by: michaelevdokimov ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@267883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	5de8e8a0b0	Merged revisions 267775 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r267775 | tilghman | 2010-06-03 20:20:17 -0500 (Thu, 03 Jun 2010) | 14 lines Merged revisions 267759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r267759 | tilghman | 2010-06-03 20:16:26 -0500 (Thu, 03 Jun 2010) | 7 lines Make the default install path appear to be /usr on Linux, instead of /usr/local. Also, reorganize the options, so that they're more alphabetical. (closes issue #17013) Reported by: klaus3000 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@267787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	6da8627ed4	Add fields back to ast_translator for ABI reasons. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@267527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	efe0f4231e	Merged revisions 267492 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r267492 | mmichelson | 2010-06-03 12:09:11 -0500 (Thu, 03 Jun 2010) | 6 lines Remove unnecessary code relating to PLC. The logic for handling generic PLC is now handled in ast_write in channel.c instead of in translation code. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@267507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	3c047b9a31	Merged revisions 267065 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r267065 | jpeeler | 2010-06-02 12:29:35 -0500 (Wed, 02 Jun 2010) | 12 lines Fix infinite loop when loading codec speex This changes the sample slinear frame data to contain non-zero data so that translation calculations for speex works when preprocessing and VAD is turned on. The encoder expects samples to be returned, but when attempted with the mentioned two options and silent sample frames everything was discarded. (closes issue #17240) Reported by: seandarcy Review: https://reviewboard.asterisk.org/r/682/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@267073 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	a199f18aa6	Merged revisions 265747 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r265747 | tilghman | 2010-05-25 19:29:40 -0500 (Tue, 25 May 2010) | 8 lines Use configure to determine the prefixes and include directories properly. This ensures cross-platform compatibility, even among Linux distributions, which don't always put headers in the same place. (closes issue #17391) Reported by: loloski ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265748 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	a2b47e4eda	Merged revisions 265320,265467 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r265320 | twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines Add the FullyBooted AMI event It is possible to connect to the manager interface before all Asterisk modules are loaded. To ensure that an application does not send AMI actions that might require a module that has not yet loaded, the application can listen for the FullyBooted manager event. It will be sent upon connection if all modules have been loaded, or as soon as loading is complete. The event: Event: FullyBooted Privilege: system,all Status: Fully Booted Review: https://reviewboard.asterisk.org/r/639/ ........ r265467 | twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line Merge the rest of the FullyBooted patch ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	d3ae642446	Merged revisions 265090 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r265090 | mmichelson | 2010-05-21 16:08:51 -0500 (Fri, 21 May 2010) | 15 lines Merged revisions 265089 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May 2010) | 8 lines Don't hang up on a queue caller if the file we attempt to play does not exist. This also fixes a documentation mistake in file.h that made my original attempt to correct this problem not work correctly. (closes issue #17061) Reported by: RoadKill ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	f83214b612	Merged revisions 265000 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r265000 | mmichelson | 2010-05-21 11:54:21 -0500 (Fri, 21 May 2010) | 9 lines Merged revisions 264999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri, 21 May 2010) | 3 lines Fix grammatical error in comment. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265001 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	9a124ea569	Merged revisions 264997 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r264997 | mmichelson | 2010-05-21 11:44:27 -0500 (Fri, 21 May 2010) | 38 lines Merged revisions 264996 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May 2010) | 32 lines Allow ast_safe_sleep to defer specific frames until after the sleep has concluded. From reviewboard Background: A Digium customer discovered a somewhat odd bug. The setup is that parties A and B are bridged, and party A places party B on hold. While party B is listening to hold music, he mashes a bunch of DTMF. Party A takes party B off hold while this is happening, but party B continues to hear hold music. I could reproduce this about 1 in 5 times. The issue: When DTMF features are enabled and a user presses keys, the channel that the DTMF is streamed to is placed in an ast_safe_sleep for 100 ms, the duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is read from the channel during the sleep, the frame is dropped. Thus the unhold indication is never made to the channel that was originally placed on hold. The fix: Originally, I discussed with Kevin possible ways of fixing the specific problem reported. However, we determined that the same type of problem could happen in other situations where ast_safe_sleep() is used. Using autoservice as a model, I modified ast_safe_sleep_conditional() to defer specific frame types so they can be re-queued once the sleep has finished. I made a common function for determining if a frame should be deferred so that there are not two identical switch blocks to maintain. Review: https://reviewboard.asterisk.org/r/674/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@264998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago