1321 Commits (656d2e043a5643c31ea3208828ca1b4d2761cc41)

Author	SHA1	Message	Date
Sean Bright	224b82262f	Resolve a segfault/bus error when we try to map memory that falls on a page ... boundary. The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the mmap'd region. The problem with this is that reading/writing to that extra byte outside of the bounds of the underlying fd causes a bus error. The real issue is that we are working with both a FILE * and the raw fd underneath it and not synchronizing between them. The code that was removed in ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping the fd. Looking at the manager code in 1.4 reveals that the FILE * in 'struct mansession' is never used except to create a temporary file that we immediately fdopen. This means we just need to write a 0 byte to the fd and everything will just work. The other branches require a call to fflush() which, while not a guaranteed fix, should reduce the likelihood of a crash. This all makes sense in my head. (closes issue ASTERISK-16460) Reported by: Ravelomanantsoa Hoby (hoby) Patches: issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@323559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	94fa98557d	Reverted part of r314607, as it can introduce a regression. ... Specifically, the security check for the "system" privilege was removed. If a user had the "call" privilege but not the "system" privilege, they would loose the ability to execute the system app and dialplan functions that run commands in a shell. This branch never used the "system" privilege for that purpose and did not need to be patched. AST-2011-006 (related to issue 0018787) Reported by: kobaz git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@315147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	9bd88702b7	Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so. ... Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action. AST-2011-005 AST-2011-006 (closes issue #18787) Reported by: kobaz (related to issue #18996) Reported by: tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@314607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	fa67a98ccc	Asterisk does not hangup a channel after endpoint hangs up. ... If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@313545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	059e1a2dec	Limit the number of unauthenticated manager sessions and also limit the time they have to authenticate. ... AST-2011-005 (closes issue #18996) Reported by: tzafrir Tested by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@312761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f6cd305277	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/1.4@312285 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	9dbd90f447	Don't dec the usecount of an eventqent then use it. ... ABE-2756 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@311201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Alec L Davis	ba3c804b06	core show locks: display ThreadID in hexadecimal ... Allow easier cross referencing of thread ID's with GDB backtraces (closes issue #18968) Reported by: alecdavis Patches: bug18968.diff.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@310779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	5e8dfa3e6b	"Caller*ID failed checksum" on Wildcard TDM2400P and TDM410 ... The last character in the caller id message is getting a framing error. The checksum is the last character in the message. A framing error in the checksum could be because: 1) The sender did not send a full stop bit. 2) The sender cut off the FSK carrier too soon. 3) The sender opted to send zero of the specified zero to 10 trailing mark bits and round-off errors in the code resulted in the code not being where it thought it was in the demodulated bit stream. Bit 8 of 'b' is set when parity error. Bit 9 of 'b' is set when framing error. Made ignore the framing and parity error bits if the errored character is the checksum. We can tolerate a framing/parity error there. The checksum character validates the message. (closes issue #18474) Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek (license 636) (with modifications) Tested by: nivek git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@310633 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	e90101cecf	Don't broadcast FullyBooted to every AMI connection ... The FullyBooted event should not be sent to every AMI connection every time someone connects via AMI. It should only be sent to the user who just connected. (closes issue #18168) Reported by: FeyFre Patches: bug0018168.patch uploaded by FeyFre (license 1142) Tested by: FeyFre, twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@308813 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	ebc809b6d4	silence gcc 4.2 compiler warning ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@308721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	2e714923c3	Properly check the bounds of arrays when decoding UDPTL packets. Also, remove broken support for receiving UDPTL packets larger than 16k. That shouldn't ever happen anyway. ... AST-2011-002 FAX-281 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@308413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	af36e55080	Remove color when executing commands via a remote console. ... Essentially this makes '-x' imply '-n' on rasterisk. This was done in a different and incomplete way previously, which I'll be reverting shortly. (issue #18776) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@307534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	9fe4bc60b5	Minor AST_FRAME_TEXT related issues. ... * Include the null terminator in the buffer length. When the frame is queued it is copied. If the null terminator is not part of the frame buffer length, the receiver could see garbage appended onto it. * Add channel lock protection with ast_sendtext(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@305888 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	13d35425dc	Get rid of unused 'verbose' field in ast_udptl ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@304242 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	b3903adbb9	Backport the Proceeding application. ... Added in trunk -r129399. Enable the workaround for issue #17085 and #18509. (issue #17085) (issue #18509) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@303747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	a4c9384154	Fix a couple of mistakes in the backport of this patch to 1.4. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@303552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	58a666ac5a	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/1.4@303546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	4717c625e9	Don't store the thread id for the manager session in the structure we pass to ... the thread for the manager session. ABE-2543 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@301591 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	4b11743a18	Fix CPU spike when pressing DTMF after agent login. ... The problem here is that DTMF was being continuously deferred and requeued since ast_safe_sleep is called in a loop. There are serveral other places in the code that sleeps and then loops in a similar fashion. Because of this fact I opted to not defer DTMF any more, which will not affect the original fix: https://reviewboard.asterisk.org/r/674 (closes issue #18130) Reported by: rgj git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@301502 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	08fcb05bc6	Prevent buffer overflows in ast_uri_encode() ... ABE-2705 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@301305 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	24118f5839	Typos: recieved => received ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@299002 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	78febf3647	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/1.4@298905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	d644f402d0	Fix reference and container leaks when running 'astobj2 test.' ... We need to make sure that ao2_iterator_destroy is called once for each time that ao2_iterator_init is called. Also make sure to unref a newly allocated object that we've linked into a container. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@298345 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	cd535276c1	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: https://origsvn.digium.com/svn/asterisk/branches/1.4@297823 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	0135ff19ed	Avoid a crash if we don't pass an argument to 'astobj2 test.' ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@297775 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	59db525cc0	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: https://origsvn.digium.com/svn/asterisk/branches/1.4@297310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	196032208f	Get rid of the annoying startup and shutdown errors on OS X. ... This mainly deals with the problem of constructors on platforms where an explicit constructor order cannot be specified (any system with gcc 4.2 or less). However, this is only a problem on those systems where we need to initialize mutexes with a constructor, because we have other code that also relies upon constructors, and we cannot specify that mutexes are initialized first (and destroyed last). There are two approaches to dealing with this issue, related to whether the code exists in the core Asterisk binary or in a separate code module. In the core case, constructors are run immediately upon load, and the file_versions list mutex needs to be already initialized, as it is referenced in the first constructor within each core source file. In this case, we use pthread_once to ensure that the mutex is initialized immediately before it is used for the first time. The only caveat is that the mutex is not ever destroyed, but because this is the core, it makes no real difference; the only time when destruction is safe would be just prior to process destruction, which takes care of that anyway. And due to using pthread_once, the mutex will never be reinitialized, which means only one structure has leaked at the end of the process. Hence, it is not a problematic leak. The second approach is to use the load_module and unload_module routines, which, for obvious reasons, exist only in loadable modules. In this second case, we don't have a problem with the constructors, but only with destructor order, because mutexes can be destroyed before their final usage is employed. However, we need the mutexes to still be destroyed, in certain scenarios: if the module is unloaded prior to the process ending, it should be clean, with no allocations by the module hanging around after that point in time. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@296867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	0876c9fbe1	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: https://origsvn.digium.com/svn/asterisk/branches/1.4@296309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	f95e212afc	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: https://origsvn.digium.com/svn/asterisk/branches/1.4@296213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	41656ad0bf	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: https://origsvn.digium.com/svn/asterisk/branches/1.4@296082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	72d3cd8b47	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: https://origsvn.digium.com/svn/asterisk/branches/1.4@296000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	74270307c9	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: https://origsvn.digium.com/svn/asterisk/branches/1.4@295906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	d32582d42f	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/1.4@295790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	b174c8f30e	Dead code elimination in channel.c:ast_channel_bridge() variable who. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@295280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	676cb0993b	I didn't mean to merge this, sorry ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@294739 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	80740e014d	One small addition to 294384 found while very carefully merging to 1.6. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@294641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	7876a359ad	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/1.4@294384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8974f3a50e	"!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: https://origsvn.digium.com/svn/asterisk/branches/1.4@293194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	1c2445e81b	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: https://origsvn.digium.com/svn/asterisk/branches/1.4@291577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	6fa0d857f5	Oops, incorrect range (although unallocated at ARIN) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@291263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	ab95a82f60	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: https://origsvn.digium.com/svn/asterisk/branches/1.4@290862 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	18d5cf58ea	Add documentation for undocumented option to AMI action originate ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@289949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	8f6ca92a9d	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/1.4@289797 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	3d130fc262	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: https://origsvn.digium.com/svn/asterisk/branches/1.4@289338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	0dab4b2303	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: https://origsvn.digium.com/svn/asterisk/branches/1.4@289177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Brett Bryant	f13325ba10	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: https://origsvn.digium.com/svn/asterisk/branches/1.4@289094 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	b94a0e14a2	Solaris compatibility fixes ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@288636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	bcd4e8787c	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: https://origsvn.digium.com/svn/asterisk/branches/1.4@288339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Brett Bryant	f9b2036d86	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: https://origsvn.digium.com/svn/asterisk/branches/1.4@288005 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago