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2011-02-21Merge changes related to AST-2011-002 and FAX-281.lmadsen1-27/+21
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16.2@308507 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-17AST-2011-001lmadsen1-14/+13
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.16.1@302148 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-10Portability issue on OpenSolaris.tilghman1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@298050 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-07Merged revisions 297823 via svnmerge from jpeeler1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010) | 12 lines Revert code that changed SSRC for DTMF. Some previous behavior was attempted to be restored, but mistakingly I did not realize that the previous behavior was incorrect. This fixes DTMF not being detected since DTMF shouldn't cause the SSRC to change. (related to issue #17404) (closes issue #18189) (closes issue #18352) Reported by: marcbou Tested by: cmbaker82 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297824 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-02Merged revisions 297310 via svnmerge from twilson1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010) | 12 lines Initialize offset for adaptive jitter buffer When the adaptive jitter buffer is enabled in sip.conf, the first frame placed in the jitter buffer fails with something like: jb_warning_output: Resyncing the jb. last_delay 0, this delay -215886466, threshold 1000, new offset 215886466 This happens because the offset is not initialized before calling jb_put(). This patch modifies jb_put_first_adaptive() to set the offset to the frame's timestamp. Review: https://reviewboard.asterisk.org/r/1041/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297311 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-29I love standards. There are so many to choose from. Except when there ↵tilghman1-1/+7
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@296533 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-26Merged revisions 296309 via svnmerge from oej1-39/+48
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11 lines Fix bugs in saying numbers using the Swedish language syntax (closes issue #18355) Reported by: oej Patch by: oej Much help from Peter Lindahl. Testing by the ClearIT team during a coffee break. Review: https://reviewboard.asterisk.org/r/1033/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296351 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Merged revisions 296213 via svnmerge from russell1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010) | 6 lines Make Asterisk less crashy. Since we might not put a new translation path on the channel, go ahead and set it to NULL right after destroying the old one to ensure we don't try to free an invalid translation path later on. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296221 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Merged revisions 296082 via svnmerge from russell1-7/+17
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010) | 12 lines Fix false reporting of an error by set_format(). In the case that the native format was able to be changed to match the new requested format, the code proceeded to attempt to build a translation path, anyway. The result would be NULL, since no translation path is necessary and resulted in this function thinking an error has occurred. This case is now specifically caught and no attempt to build a translation path is attempted. Thanks to our automated tests and bamboo.asterisk.org for catching this problem and making a whole lot of noise when things started failing. :-) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296083 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Merged revisions 296000 via svnmerge from russell1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines Handle failures building translation paths more effectively. The problem scenario occurred on a heavily loaded system that was using the codec_dahdi module and exceeded the hardware transcoding capacity. The failure mode at that point was not good. The report came in to us as an Asterisk lock-up. The "core show locks" shows a ton of threads locked up (but no obvious deadlock). Upon deeper investigation, when the system is in this state, the CPU was maxed out. The CPU was being consumed by the Asterisk logger spewing messages on every audio frame for calls set up after transcoder capacity was reached. The purpose of this patch is to make Asterisk handle failures to create a translation path in a more graceful manner. If we can't translate, then the call just needs to be dropped, as it's not going to work. These are the changes: 1) In set_format() of channel.c (which is called by set_read_format() and set_write_format()), it was ignoring if ast_translator_build_path() failed and returned NULL. It now pays attention to that case and returns a result reflecting failure. With this change in place, the bridging code will immediately detect a failure and end the bridge instead of proceeding to try to bridge frames that can't be translated and making channel drivers freak out by sending them frames in a format they weren't expecting. 2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was ignored. It is now reflected in the return value of the function. This didn't turn out to have any affect on the bug, but seemed like a good change to leave in. 3) In app_dial(), when only sending a call to a single endpoint, it will attempt to do some bridging of its own of early audio. It uses make_compatible() when it's going to do this. However, it ignored failure from make compatible. So, even with the fix from #1, if there was early audio going through app_dial, there would still be a period of invalid frames passing through. After detecting failure here, Dial() exits. ABE-2658 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296001 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-23Merged revisions 295906 via svnmerge from oej1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8 lines Fix support of saynumber(1,n) in the Swedish language (closes issue #18353) Reported by: oej Review: https://reviewboard.asterisk.org/r/1031/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@295907 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-22Merged revisions 295790 via svnmerge from rmudgett2-44/+92
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@295843 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-20Fix cache of device state changes for multiple servers.russell1-3/+154
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@295710 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-16Merged revisions 295280 via svnmerge from rmudgett1-5/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16 Nov 2010) | 1 line Dead code elimination in channel.c:ast_channel_bridge() variable who. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@295281 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-11Merged revisions 294384 via svnmerge from jpeeler2-108/+186
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@294639 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-10Actually pay attention to documented settings in features.conf.tilghman1-0/+11
(closes issue #16757) Reported by: voxter Patches: 20101012__issue16757.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/994/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@294571 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-10Improve a debug message to be more readable and consistent.russell1-1/+2
(closes issue #18282) Reported by: klaus3000 Patches: ast_devstate2str-patch.txt uploaded by klaus3000 (license 65) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@294500 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-08Fix playback failure when using IAX with the timerfd module.jpeeler2-1/+8
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@294277 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-28Merged revisions 293194 via svnmerge from tilghman2-251/+384
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) | 5 lines "!00" evaluated as false, which is incorrect. Fixing. Reported (though the reporter did not understand he was reporting a bug) on the asterisk-users list: http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@293196 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-28Merged revisions 293194 via svnmerge from tilghman1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) | 5 lines "!00" evaluated as false, which is incorrect. Fixing. Reported (though the reporter did not understand he was reporting a bug) on the asterisk-users list: http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@293195 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Merged revisions 291577 via svnmerge from twilson1-1/+12
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010) | 21 lines Don't ignore frames that have been queued when softhangup'd When an outgoing call is answered and hung up by the far end *very* quickly, we may not read any frames and therefor end up with a call that displays the wrong disposition/DIALSTATUS. The reason is because ast_queue_hangup() immediately sets the _softhangup flag on the channel and then queues the HANGUP control frame, but __ast_read refuses to read any frames if ast_check_hangup() indicates that a hangup request has been made (which it will if _softhangup is set). So, we end up losing control frames. This change makes __ast_read continue to read frames even if a soft hangup has been requested. It queues a hangup frame to make sure that __ast_read() will still eventually return NULL. Much thanks to David Vossel for all of the reviews, discussion, and help! (closes issue #16946) Reported by: davidw Review: https://reviewboard.asterisk.org/r/740/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@291580 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-12Merged revisions 291263 via svnmerge from tilghman1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12 Oct 2010) | 2 lines Oops, incorrect range (although unallocated at ARIN) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@291264 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Fixed infinite loop in verbose/debug message output.rmudgett1-17/+38
Setting the module/filename specific message level and then changing it resulted in the linked list being looped on itself. Traversing this linked list is an infinite loop if what you are looking for is not in the list. Also plugged some CLI parsing holes in the associated CLI command: * Removing a nonexistent module from the list actually added it with a level of zero. * Setting the non-module specific level to zero is now equivalent to setting it to "off" as documented. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@291073 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-08Merged revisions 290862 via svnmerge from jpeeler1-4/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010) | 9 lines Ensure editline cleanup occurs when Ctrl-C is pressed at control console. A recent change was made to avoid a race condition on shutdown which only called the end functions from the console thread. However, when pressing Ctrl-C the quit handler is called from the signal handler thread. (closes issue #17698) Reported by: jmls ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@290863 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-07Don't crash when Set() is called without a value.russell1-2/+9
Review: https://reviewboard.asterisk.org/r/949/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@290712 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-06Allow streaming audio from a pipe.tilghman1-3/+5
(closes issue #18001) Reported by: jamicque Patches: 20100926__issue18001.diff.txt uploaded by tilghman (license 14) Tested by: jamicque git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@290575 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-04Change new pattern matcher to regard dashes the same as the old pattern ↵tilghman1-0/+4
matcher -- as visual candy to be ignored. Also change the AEL parser to not generate dashes within extensions, as those dashes would be ignored. Update the AEL tests to match this behavior. (closes issue #17366) Reported by: murf Patches: 20100727__issue17366.diff.txt uploaded by tilghman (license 14) Tested by: tilghman git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@290254 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-02Merged revisions 289949 via svnmerge from oej1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2 lines Add documentation for undocumented option to AMI action originate ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@289950 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01Merged revisions 289797 via svnmerge from jpeeler1-1/+13
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@289798 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-29Merged revisions 289338 via svnmerge from qwell2-3/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) | 8 lines Allow a manager originate to succeed on forwarded devices. The timeout to wait for an answer was being set to 0 when a device forwarded to another extension. We don't always need the timeout set like this, so make it an optional parameter, and don't use it in this case. ABE-2544 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@289339 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-29Merged revisions 289177 via svnmerge from mnicholson1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep 2010) | 8 lines Set the caller id on CDRs when it is set on the parent channel. (closes issue #17569) Reported by: tbelder Patches: 17569.diff uploaded by tbelder (license 618) Tested by: tbelder ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@289178 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-28Merged revisions 289094 via svnmerge from bbryant1-3/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010) | 14 lines Fixes an issue with the Newchannel AMI event during the Masquerading process. Fixes an issue with the Newchannel AMI event during the Masquerading process, where no Newchannel AMI event was generated for the psuedo channel used during the masquerading process. (closes issue #17987) Reported by: RadicAlish Patches: newchannel.patch.txt uploaded by RadicAlish (license 1122) Tested by: RadicAlish Review: https://reviewboard.asterisk.org/r/937/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@289095 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-24Merged revisions 288636 via svnmerge from tilghman1-0/+13
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@288637 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-22Merged revisions 288339 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010) | 11 lines Fix a 100% CPU consumption problem when setting console=yes in asterisk.conf. The handling of -c and console=yes should be the same, but they were not. When you specify -c, it sets both a flag for console module and for asterisk not to fork() off into the background. The handling of console=yes only set console mode, so you would end up with a background process() trying to run the Asterisk console and freaking out since it didn't have anything to read input from. Thanks to beagles for reporting and helping debug the problem! ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@288340 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Merged revisions 288005 via svnmerge from bbryant1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010) | 8 lines Add a check to fix a rare segmentation fault you'd get if ast_frdup couldn't allocate memory on the first frame being queued in ast_queue_frame. (closes issue #17882) Reported by: seanbright Tested by: seanbright ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@288006 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Merged revisions 287933 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21 Sep 2010) | 2 lines Less than zero is an error, not any non-zero value. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@287934 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-20ast_channel_masquerade: Avoid recursive masquerades.alecdavis1-7/+15
Check all 4 combinations of (original/clonechan) * (masq/masqr). Initially original->masq and clonechan->masqr were only checked. It's possible with multiple masq's planned - and not yet executed, that the 'original' chan could already have another masq'd into it - thus original->masqr would be set, that masqr would lost. Likewise for the clonechan->masq. (closes issue #16057;#17363) Reported by: amorsen;davidw,alecdavis Patches: based on bug16057.diff4.txt uploaded by alecdavis (license 585) Tested by: ramonpeek, davidw, alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@287685 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-20Use ast_str when processing hint state changesmnicholson1-6/+10
Merged revisions 287555 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep 2010) | 5 lines Use ast_dynamic_str when processing hint state changes (related to issue #17928) Reported by: mdu113 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@287558 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-19Merged revisions 287469 via svnmerge from oej1-1/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7 lines Make sure we always free variables properly in manager originate. (closes issue #17891) reported, solved and tested by oej Review: https://reviewboard.asterisk.org/r/869/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@287470 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-17Merged revisions 287307 via svnmerge from mnicholson1-2/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep 2010) | 5 lines Use ast_strdup() instead of ast_strdupa() while processing in ast_hint_state_changed(). (related to issue #17928) Reported by: mdu113 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@287308 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-16Merged revisions 287118 via svnmerge from mnicholson1-3/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep 2010) | 8 lines Don't limit hint processing in ast_hint_state_changed() to AST_MAX_EXTENSION length strings. (closes issue #17928) Reported by: mdu113 Patches: 20100831__issue17928.diff.txt uploaded by tilghman (license 14) Tested by: mdu113 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@287119 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-16Merged revisions 287114 via svnmerge from mnicholson1-10/+10
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep 2010) | 8 lines Don't stop printing cdr variables if we encounter one with a blank name or value. (closes issue #17900) Reported by: under Patches: core-show-channel-cdr-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@287115 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-14Merged revisions 286679 via svnmerge from mnicholson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep 2010) | 7 lines Only drop duplicate answer frames if the channel is bridged. Back in r3710 ast_read() was modified to drop answer frames on channels that were in the UP state. This modification prevented bridges that were up before the answer from being broken and reestablished by an ANSWER control frame. That change also prevents pickup of channels called from the ast_dial framework from working properly. The ast_dial framework expects to see an ANSWER frame after dialing and the pickup code queues one but ast_read() drops it. This new change only drops ANSWER frames when the channel is bridged, allowing the answer queued by the pickup code to properly pass through ast_read() on to the ast_dial framework. ABE-2473 (related to issue #2342) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286681 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-13C precedence got metilghman1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286557 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-13Refactor conversion to ast_poll() to fix callparking regression.tilghman1-21/+23
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286527 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-11Merged revisions 286267 via svnmerge from oej1-1/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4 lines Handle error response when we can't make file compatible Review: https://reviewboard.asterisk.org/r/911/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286268 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 286023 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10 Sep 2010) | 2 lines Missing newline ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286024 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-09Merged revisions 285742 via svnmerge from qwell1-1/+13
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) | 9 lines Transmit silence when reading DTMF in ast_readstring. Otherwise, you could get issues with DTMF timeouts causing hangups. (closes issue #17370) Reported by: makoto Patches: channel-readstring-silence-generator.patch uploaded by makoto (license 38) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@285744 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-09Fixes an issue with dialplan pattern matching where the specificity for ↵bbryant9-188/+286
pattern ranges and pattern special characters was inconsistent. (closes issue #16903) Reported by: Nick_Lewis Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@285710 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29Merged revisions 280448 via svnmerge from dvossel1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010) | 12 lines fixes issue with translator frame not getting freed A translator frame even if it local storage so the translation path can be freed. This issue prevented g729 licenses from being freed up. (closes issue #17630) Reported by: manvirr Patches: encoder_fix.diff uploaded by dvossel (license 671) Tested by: manvirr, dvossel ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280449 f38db490-d61c-443f-a65b-d21fe96a405b
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