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2009-03-03Ensure chan->fdno always gets reset to -1 after handling a channel fd event.russell1-1/+4
Since setting fdno to -1 had to be moved, a couple of other code paths that do process an fd event return early and do not pass through the code path where it was moved to. So, set it to -1 in a few other places, too. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@179741 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Move where fdno is set to the default value to *after* the read callback of ↵file1-6/+6
the channel driver is called. We have to do this as the underlying channel driver may need the fdno value to determine what to read. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@179671 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Make it easier to detect an improper call to ast_read().russell1-0/+12
When you call ast_waitfor() on a channel, the index into the channel fds array that holds the file descriptor that poll() determines has input available is stored in fdno. This patch clears out this value after a call to ast_read() and also reports errors if ast_read() is called without an fdno set. From a discussion on the asterisk-dev list. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@179608 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Fix bridging regression from commit 176701jpeeler1-1/+1
This fixes a bad regression where the bridge would exit after an attended transfer was made. The problem was due to nexteventts getting set after the masquerade which caused the bridge to return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: tim_ringenbach git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@179536 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Ensure that only one thread is calling ast_settimeout() on a channel at a time.russell1-0/+2
For example, with an IAX2 channel, you can have both the channel thread and the chan_iax2 processing threads calling this function, and doing so twice at the same time is a bad thing. (Found in a debugging session with dvossel and mmichelson) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@179461 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Modify bridging to properly evaluate DTMF after first warning is playedjpeeler1-14/+22
The main problem is currently if the Dial flag L is used with a warning sound, DTMF is not evaluated after the first warning sound. To fix this, a flag has been added in ast_generic_bridge for playing the warning which ensures that if a scheduled warning is missed, multiple warrnings are not played back (due to a feature evaluation or waiting for digits). ast_channel_bridge was modified to store the nexteventts in the ast_bridge_config structure as that information was lost every time ast_channel_bridge was reentered, causing a hangup due to incorrect time calculations. (closes issue #14315) Reported by: tim_ringenbach Reviewed on reviewboard: http://reviewboard.digium.com/r/163/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@176701 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-23When a channel is answered make sure any indications currently playing stop. ↵file1-0/+1
Usually the phone would do this but if the channel was already answered then they are being generated by Asterisk and we darn well need to stop them. (closes issue #14249) Reported by: RadicAlish git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@170648 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-23Fix broken call pickupmmichelson1-1/+1
There was a subtle change in ast_do_masquerade which resulted in failed attempts to pickup calls. The problem was that the value of the AST_FLAG_OUTGOING flag was copied from the clone to the original channel. In the case of call pickup, this meant that the AST_FLAG_OUTGOING flag ended up being cleared on the channel that was attempting to execute the pickup. Because this flag was not set, when ast_read came across an answer frame, it ignored it. The result of this was that the calling channel was never properly answered. This fix changes the behavior in ast_do_masquerade to set the flags on the original channel to the union of the flags on the clone channel. This way, if the AST_FLAG_OUTGOING flag is set on either of the two channels involved in the masquerade, the resulting channel will have the flag set as well. (closes issue #14206) Reported by: francesco_r Patches: 14206.patch uploaded by putnopvut (license 60) Tested by: francesco_r, aragon, putnopvut git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@170392 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-13Revert unnecessary indications API change from rev 122314russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@168561 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-23Fix a crash resulting from a datastore with inheritance but no duplicate ↵mmichelson1-1/+1
callback The fix for this is to simply set the newly created datastore's data pointer to NULL if it is inherited but has no duplicate callback. (closes issue #14113) Reported by: francesco_r Patches: 14113.patch uploaded by putnopvut (license 60) Tested by: francesco_r git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@166568 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-23Use the integer form of condition for integer comparisons.tilghman1-1/+4
(closes issue #14127) Reported by: andrew git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@166509 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-19Backport of AUDIOHOOK_INHERIT for Asterisk 1.4mmichelson1-2/+2
(closes issue #13538) Reported by: mbit Patches: 13538.patch uploaded by putnopvut (license 60) Tested by: putnopvut git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@166157 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-19(closes issue #13480)jpeeler1-5/+1
Reported by: tzafrir Replace a bunch of if defined checks for Zaptel/DAHDI through several new defines in dahdi_compat.h. This removes a lot of code duplication. Example from bug: #ifdef HAVE_ZAPTEL fd = open("/dev/zap/pseudo", O_RDWR); #else fd = open("/dev/dahdi/pseudo", O_RDWR); #endif is replaced with: fd = open(DAHDI_FILE_PSEUDO, O_RDRW); git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@165991 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-15Handle a case where a call can be bridged to a channel that is still ringing.russell1-50/+119
The issue that was reported was about a case where a RINGING channel got redirected to an extension to pick up a call from parking. Once the parked call got taken out of parking, it heard silence until the other side answered. Ideally, the caller that was parked would get a ringing indication. This patch fixes this case so that the caller receives ringback once it comes out of parking until the other side answers. The fixes are: - Make sure we remember that a channel was an outgoing channel when doing a masquerade. This prevents an erroneous ast_answer() call on the channel, which causes a bogus 200 OK to be sent in the case of SIP. - Add some additional comments to explain related parts of code. - Update the handling of the ast_channel visible_indication field. Storing values that are not stateful is pointless. Control frames that are events or commands should be ignored. - When a bridge first starts, check to see if the peer channel needs to be given ringing indication because the calling side is still ringing. - Rework ast_indicate_data() a bit for the sake of readability. (closes issue #13747) Reported by: davidw Tested by: russell Review: http://reviewboard.digium.com/r/90/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@164201 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-12Resolve issues that could cause DTMF to be processed out of order.russell1-41/+92
These changes come from team/russell/issue_12658 1) Change autoservice to put digits on the head of the channel's frame readq instead of the tail. If there were frames on the readq that autoservice had not yet read, the previous code would have resulted in out of order processing. This required a new API call to queue a frame to the head of the queue instead of the tail. 2) Change up the processing of DTMF in ast_read(). Some of the problems were the result of having two sources of pending DTMF frames. There was the dtmfq and the more generic readq. Both were used for pending DTMF in various scenarios. Simplifying things to only use the frame readq avoids some of the problems. 3) Fix a bug where a DTMF END frame could get passed through when it shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation, and a digit arrived before emulation was complete, digits would get processed out of order. (closes issue #12658) Reported by: dimas Tested by: russell, file Review: http://reviewboard.digium.com/r/85/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@163448 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-05This fix was prompted by communication from user, who was seeing thousands ↵murf1-1/+2
of error logs... looks like EAGAIN. Made such uninteresting. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@154685 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-01fix a bunch of potential problems found by gcc 4.3.x, primarily bare strings ↵kpfleming1-4/+12
being passed to printf()-like functions and ignored results from read()/write() and friends git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@153337 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-17Correctly allow chan_dahdi to compile against older versions of Zaptel.qwell1-1/+1
Don't always define HAVE_ZAPTEL_CHANALARMS (since we check if it's defined..) Minor cleanup to make things clear. (closes issue #13726) Reported by: tzafrir Patches: dahdi_def.diff uploaded by tzafrir (license 46) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@150557 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-05A small change to prevent double-posting of CDR's; thanks to Daniel Ferrer ↵murf1-1/+1
for bringing it to our attention git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@141156 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-02After reconsidering, with respect to 13409, ast_cdr_detach should be OK, ↵murf1-1/+1
better in fact, than ast_cdr_free, which generates lots of useless warnings that will undoubtably generate complaints. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@140690 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-02(closes issue #13409)murf1-0/+6
Reported by: tomaso Patches: asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license 564) I basically spent the day, verifying that this patch solves the problem, and doesn't hurt in non-problem cases. Why valgrind did not plainly reveal this leak absolutely mystifies and stuns me. Many, many thanks to tomaso for finding and providing the fix. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@140670 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-06Fix a longstanding bug in channel walking logic, and fix the explanation totilghman1-10/+13
make sense. (Closes issue #13124) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@135949 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-06Merging the issue11259 branch.mmichelson1-0/+5
The purpose of this branch was to take into account "burps" which could cause jitterbuffers to misbehave. One such example is if the L option to Dial() were used to inject audio into a bridged conversation at regular intervals. Since the audio here was not passed through the jitterbuffer, it would cause a gap in the jitterbuffer's timestamps which would cause a frames to be dropped for a brief period. Now ast_generic_bridge will empty and reset the jitterbuffer each time it is called. This causes injected audio to be handled properly. ast_generic_bridge also will empty and reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE frame since the change in audio source could negatively affect the jitterbuffer. All of this was made possible by adding a new public API call to the abstract_jb called ast_jb_empty_and_reset. (closes issue #11259) Reported by: plack Tested by: putnopvut git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@135841 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-05(closes issue #12982)murf1-1/+6
Reported by: bcnit Tested by: murf I discovered that also, in the previous bug fixes and changes, the cdr.conf 'unanswered' option is not being obeyed, so I fixed this. And, yes, there are two 'answer' times involved in this scenario, and I would agree with you, that the first answer time is the time that should appear in the CDR. (the second 'answer' time is the time that the bridge was begun). I made the necessary adjustments, recording the first answer time into the peer cdr, and then using that to override the bridge cdr's value. To get the 'unanswered' CDRs to appear, I purposely output them, using the dial cmd to mark them as DIALED (with a new flag), and outputting them if they bear that flag, and you are in the right mode. I also corrected one small mention of the Zap device to equally consider the dahdi device. I heavily tested 10-sec-wait macros in dial, and without the macro call; I tested hangups while the macro was running vs. letting the macro complete and the bridge form. Looks OK. Removed all the instrumentation and debug. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@135799 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-25Fix some errant device states by making the devicestate API more strict intilghman1-2/+11
terms of the device argument (only without the unique identifier appended). (closes issue #12771) Reported by: davidw Patches: 20080717__bug12771.diff.txt uploaded by Corydon76 (license 14) Tested by: davidw, jvandal, murf git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@133649 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-11a whole pile of Zaptel/DAHDI compatibility work, with lots more to come... ↵kpfleming1-5/+0
this tree is not yet ready for users to be easily upgrading or switching, but it needs to be :-) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@130298 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-03The CDRfix4/5/6 omnibus cdr fixes.murf1-33/+8
(closes issue #10927) Reported by: murf Tested by: murf, deeperror (closes issue #12907) Reported by: falves11 Tested by: murf, falves11 (closes issue #11849) Reported by: greyvoip As to 11849, I think these changes fix the core problems brought up in that bug, but perhaps not the more global problems created by the limitations of CDR's themselves not being oriented around transfers. Reopen if necc, but bug reports are not the best medium for enhancement discussions. We need to start a second-generation CDR standardization effort to cover transfers. (closes issue #11093) Reported by: rossbeer Tested by: greyvoip, murf git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@127663 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-19Change informative messages to use the _multiple variant when multiple formatstilghman1-4/+6
are possible. (Closes issue #12848) Reported by klaus3000 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@123930 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-12Adds DAHDI support alongside Zaptel. DAHDI usage favored, but all Zap stuff ↵jpeeler1-17/+28
should continue working. Release announcement to follow. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@122314 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-12Occasionally, the alertpipe loses its nonblocking status, so detect and correcttilghman1-3/+26
that situation before it causes a deadlock. (Reported and tested by ctooley via #asterisk-dev) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@122130 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-11Make calls to ast_assert() actually test something, so that the error messagetilghman1-2/+2
printed is not nonsensical (reported by mvanbaak via #asterisk-bugs). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@121861 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-10Update BRIDGEPEER variable before we do a generic bridge in case we just ↵file1-0/+6
broke out of a native bridge and fell through to generic. (closes issue #12815) Reported by: ramonpeek git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@121442 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-09Do not attempt to do emulation if an END digit is received and the length isrussell1-1/+4
less than the defined minimum digit length, and the other end only wants END digits (SIP INFO, for example). (closes issue #12778) Reported by: tsearle Patches: 12778.rev1.txt uploaded by russell (license 2) Tested by: tsearle git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@121280 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-14Add ast_assert(), which can be used to handle fatal errors. It is only compiledrussell1-2/+2
in if dev-mode is enabled, and only aborts if DO_CRASH is defined. (inspired by issue #12650) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@116463 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-13A change to the way channel locks are handled when DEBUG_CHANNEL_LOCKS is ↵mmichelson1-5/+15
defined. After debugging a deadlock, it was noticed that when DEBUG_CHANNEL_LOCKS is enabled in menuselect, the actual origin of channel locks is obscured by the fact that all channel locks appear to happen in the function ast_channel_lock(). This code change redefines ast_channel_lock to be a macro which maps to __ast_channel_lock(), which then relays the proper file name, line number, and function name information to the core lock functions so that this information will be displayed in the case that there is some sort of locking error or core show locks is issued. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@116088 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-22I thought I was going to be able to leave 1.4 alone, but that was not the case.russell1-2/+12
I ran into some problems with G.722 in 1.4, so I have merged in all of the fixes in this area that I have made in trunk/1.6.0, and things are happy again. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114550 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-14Increase the retry count when attempting to show channels. This apparentlymmichelson1-2/+2
cleared an issue someone was seeing when attempting to show channels when the load was high. (closes issue #11667) Reported by: falves11 Patches: 11677.txt uploaded by russell (license 2) Tested by: falves11 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114117 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-14Save a local copy of the generate callback prior to unlocking the channel inmmichelson1-1/+5
case the generate callback goes NULL on us after the channel is unlocked. Thanks to Russell for pointing this need out to me. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@114106 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-07This fix prevents a deadlock that was experienced in chan_local. There wasmmichelson1-0/+10
deadlock prevention in place in chan_local, but it would not work in a specific case because the channel was recursively locked. By unlocking the channel prior to calling the generator's generate callback in ast_read_generator_actions(), we prevent the recursive locking, and therefore the deadlock. (closes issue #12307) Reported by: callguy Patches: 12307.patch uploaded by putnopvut (license 60) Tested by: callguy git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@113065 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-13Fix another issue that was causing crashes in chanspy. This introduces a newrussell1-1/+7
datastore callback, called chan_fixup(). The concept is exactly like the fixup callback that is used in the channel technology interface. This callback gets called when the owning channel changes due to a masquerade. Before this was introduced, if a masquerade happened on a channel being spyed on, the channel pointer in the datastore became invalid. (closes issue #12187) (reported by, and lots of testing from atis) (props to file for the help with ideas) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@108583 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-12(closes issue #12187, reported by atis, fixed by me after some brainstormingrussell1-8/+13
on the issue with mmichelson) - Update copyright info on app_chanspy. - Fix a race condition that caused app_chanspy to crash. The issue was that the chanspy datastore magic that was used to ensure that spyee channels did not disappear out from under the code did not completely solve the problem. It was actually possible for chanspy to acquire a channel reference out of its datastore to a channel that was in the middle of being destroyed. That was because datastore destruction in ast_channel_free() was done near the end. So, this left the code in app_chanspy accessing a channel that was partially, or completely invalid because it was in the process of being free'd by another thread. The following sort of shows the code path where the race occurred: ============================================================================= Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy) --------------------------------------||------------------------------------- ast_channel_free() || - remove channel from channel list || - lock/unlock the channel to ensure || that no references retrieved from || the channel list exist. || --------------------------------------||------------------------------------- || channel_spy() - destroy some channel data || - Lock chanspy datastore || - Retrieve reference to channel || - lock channel || - Unlock chanspy datastore --------------------------------------||------------------------------------- - destroy channel datastores || - call chanspy datastore d'tor || which NULL's out the ds' || - Operate on the channel ... reference to the channel || || - free the channel || || || - unlock the channel --------------------------------------||------------------------------------- ============================================================================= git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@108135 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-12Destroy the channel lock after the channel datastores.russell1-1/+2
(inspired by issue #12187) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@108031 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-10Resolve a compiler warning.russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@107102 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-10Fix a race condition where the generator can go awayrussell1-1/+5
(closes issue #12175, reported by edantie, patched by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@107099 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07Ignore source update control frame.file1-0/+1
(closes issue #12168) Reported by: plack git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@106788 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07Safely use the strncat() function.tilghman1-2/+2
(closes issue #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt uploaded by Corydon76 (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@106552 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-05Add a control frame to indicate the source of media has changed. Depending ↵file1-0/+12
on the underlying technology it may need to change some things. (closes issue #12148) Reported by: jcomellas git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@106235 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-03Merge in some changes from team/russell/autoservice-nochans-1.4russell1-5/+6
These changes fix up some dubious code that I came across while auditing what happens in the autoservice thread when there are no channels currently in autoservice. 1) Change it so that autoservice thread doesn't keep looping around calling ast_waitfor_n() on 0 channels twice a second. Instead, use a thread condition so that the thread properly goes to sleep and does not wake up until a channel is put into autoservice. This actually fixes an interesting bug, as well. If the autoservice thread is already running (almost always is the case), then when the thread goes from having 0 channels to have 1 channel to autoservice, that channel would have to wait for up to 1/2 of a second to have the first frame read from it. 2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no channels and no fds to poll() on, such as was the case with the previous code for the autoservice thread. In this case, the code would call alloca(0), and pass the result as the first argument to poll(). In this case, the 2nd argument to poll() specified that there were no fds, so this invalid pointer shouldn't actually get dereferenced, but, this code makes it explicit and ensures the pointers are NULL unless we have valid data to put there. (related to issue #12116) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@105563 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-03It is possible for no audio to pass between the current digit and next digit ↵file1-1/+8
so expand logic that clears emulation to AST_FRAME_NULL. (closes issue #11911) Reported by: edgreenberg Patches: v1-11911.patch uploaded by dimas (license 88) Tested by: tbsky git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@105560 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-18Ensure that emulated DTMFs do not get interrupted by another begin frame.file1-1/+1
(closes issue #11740) Reported by: gserra Patches: v1-11740.patch uploaded by dimas (license 88) (closes issue #11955) Reported by: tsearle (closes issue #10530) Reported by: xmarksthespot git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@103801 f38db490-d61c-443f-a65b-d21fe96a405b