Age | Commit message (Collapse) | Author |
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Reported by: pj
Two of the three places ast_waitfor_nandfds could branch off to did not clear outfd and exception. If the calling function did not clear these there was a chance they could get a false positive on testing to see whether they were set.
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unless the outfd is initialized to -1 before calling the nandfds func
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r79099 | murf | 2007-08-10 14:53:43 -0600 (Fri, 10 Aug 2007) | 1 line
From a user complaint on #asterisk, I have forced pbx_spool to explain what reason codes mean, when they are logged
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scalability and is done in such a way that we should be able to add support for other poll() replacements.
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listen and manipulate the audio going through a channel.
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of the DTMF digit(s) to be specified and make the SendDTMF application have the capability to use it.
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r78275 | file | 2007-08-06 18:41:13 -0300 (Mon, 06 Aug 2007) | 2 lines
Add additional DTMF log messages to help when debugging issues.
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Being bitmasks, it is a lot easier to read this way.
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r77785 | russell | 2007-07-30 13:55:15 -0500 (Mon, 30 Jul 2007) | 3 lines
file and I both committed changes for issue #10301. Remove a duplicated
assignment to restore the original value of the previous channel.
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r77780 | russell | 2007-07-30 12:29:43 -0500 (Mon, 30 Jul 2007) | 16 lines
(closes issue #10301)
Reported by: fnordian
Patches:
asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110)
Additional changes by me
Fix some problems in channel_find_locked() which can cause an infinite loop.
The reference to the previous channel is set to NULL in some cases. These changes
ensure that the reference to the previous channel gets restored before needing
it again.
I'm not convinced that the code that is setting it to NULL is really the right
thing to do. However, I am making these changes to fix the obvious problem
and just leaving an XXX comment that it needs a better explanation that what
is there now.
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r77771 | file | 2007-07-30 12:47:52 -0300 (Mon, 30 Jul 2007) | 6 lines
(closes issue #10301)
Reported by: fnordian
Patches:
asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110)
Restore previous behavior where if we failed to lock the channel we wanted we would return to exactly the same point as if we had just reentered the function.
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r77460 | file | 2007-07-26 20:19:04 -0300 (Thu, 26 Jul 2007) | 4 lines
(closes issue #10302)
Reported by: litnialex
If a DTMF end frame comes from a channel without a begin and it is going to a technology that only accepts end frames (aka INFO) then use the minimum DTMF duration if one is not in the frame already.
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(closes issue #10277, patches by mvanbaak)
Basically, this changes ...
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3, "Something\n");
to ...
ast_verb(3, "Something\n");
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r77154 | mmichelson | 2007-07-25 16:52:47 -0500 (Wed, 25 Jul 2007) | 3 lines
chan->emulate_dtmf_duration is an unsigned int, not a signed int, so use %u instead of %d in the format string
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r76132 | russell | 2007-07-20 13:22:24 -0500 (Fri, 20 Jul 2007) | 6 lines
Use the define that specifies the default length of an artificially created
DTMF digit in the ast_senddigit() function. The define is set to 100ms by
default, which is the same thing that this function was using. But, using
the define lets changes take effect in this case, as well as the others where
it was already used.
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universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
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where uint64_t is NOT acceptable for %lld, and also works on 32-bit machines. At least, with gcc.
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run on peer. Had to upgrade ast_flag stuff to 64 bits to do this.
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r74922 | file | 2007-07-12 16:17:59 -0300 (Thu, 12 Jul 2007) | 2 lines
Whoops... didn't want this to be returned to 0 each iteration.
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r74888 | file | 2007-07-12 14:16:28 -0300 (Thu, 12 Jul 2007) | 2 lines
When waiting for a digit ensure that a begin frame was received with it, not just an end frame. (issue #10084 reported by rushowr)
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r73355 | file | 2007-07-05 11:21:44 -0300 (Thu, 05 Jul 2007) | 10 lines
Merged revisions 73349 via svnmerge from
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r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2 lines
Tweak spy locking. (issue #9951 reported by welles)
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r72888 | file | 2007-07-02 11:32:59 -0300 (Mon, 02 Jul 2007) | 2 lines
Added additional DTMF debug messages for when emulation occurs.
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r72257 | file | 2007-06-27 16:25:24 -0400 (Wed, 27 Jun 2007) | 10 lines
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r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2 lines
I may possibly get shot for doing this... but... defer CDR processing until after the channel has been dealt with. This should eliminate all of the issues with channels going funky (SIP/PRI) when you are posting CDRs to a database that is either slow or unavailable and do not want to enable batching.
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r72148 | file | 2007-06-27 13:31:50 -0400 (Wed, 27 Jun 2007) | 2 lines
Make the ast_read_noaudio API call behave better under circumstances where DTMF emulation was happening and a generator was setup. (issue #10065 reported by stevefeinstein)
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r70062 | murf | 2007-06-19 12:23:23 -0600 (Tue, 19 Jun 2007) | 9 lines
Merged revisions 70053 via svnmerge from
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r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1 line
This fixes 9246, where channel variables are not available in the 'h' exten, on a 'ZOMBIE' channel. The fix is to consolidate the channel variables during a masquerade, and then copy the merged variables back onto the clone, so the zombie has the same vars that the 'original' has.
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r69987 | file | 2007-06-19 12:24:31 -0400 (Tue, 19 Jun 2007) | 10 lines
Merged revisions 69986 via svnmerge from
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r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2 lines
Update BRIDGEPEER variable if set to the new channel name when a masquerade happens. (issue #9699 reported by dimas)
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(issue #9983, eliel)
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(issue #9957, patches from mvanbaak, caio1982, critch, and dimas)
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r69010 | russell | 2007-06-12 14:13:41 -0500 (Tue, 12 Jun 2007) | 12 lines
In ast_channel_make_compatible(), just return if the channels' read and write
formats already match up. There are code paths that call this function on a
pair of channels multiple times. This made calls fail that were using g729
in some cases. The reason is that codec_g729a will unregister itself from the
list of available translators will all licenses are in use. So, the first
time the function got called, the right translation path was allocated.
However, the second time it got called, the code would not find a translation
path to/from g729 and make the call fail, even if the channel actually already
had a g729 translation path allocated.
(SPD-32)
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r68683 | file | 2007-06-11 10:33:12 -0400 (Mon, 11 Jun 2007) | 10 lines
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r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2 lines
Improve deadlock handling of the channel list. (issue #8376 reported by one47)
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r68157 | file | 2007-06-07 14:39:52 -0400 (Thu, 07 Jun 2007) | 2 lines
Fix logic when doing a name based channel search for a structure when you want to start from a specific point in the channel list. (issue #9324 reported by slavon)
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guidelines changes
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r67716 | russell | 2007-06-06 11:55:59 -0500 (Wed, 06 Jun 2007) | 13 lines
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r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) | 5 lines
We have some bug reports showing crashes due to a double free of a channel.
Add a sanity check to ast_channel_free() to make sure we don't go on trying
to free a channel that wasn't found in the channel list.
(issue #8850, and others...)
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r66076 | russell | 2007-05-24 17:23:59 -0500 (Thu, 24 May 2007) | 1 line
if the string field init fails, clean up the stuff that was allocated already
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r66070 | russell | 2007-05-24 17:07:39 -0500 (Thu, 24 May 2007) | 2 lines
Check the result of ast_string_field_init() in ast_channel_alloc()
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class in musiconhold.conf, a caller on hold may press this digit to switch
to listening to that music class.
This involved adding a new callback for generators, which allow generators
to get notified of DTMF from the channel they are running on. Then, a callback
was implemented for the music on hold generators.
(patch from bbryant)
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r64240 | file | 2007-05-14 13:23:51 -0400 (Mon, 14 May 2007) | 2 lines
Fix scenario where if a phone that simply called Echo() put itself on hold it could never get off hold.
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r64157 | oej | 2007-05-14 12:39:12 +0200 (Mon, 14 May 2007) | 2 lines
Add hangupcause when we lack codecs for transcoding
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r63698 | file | 2007-05-09 15:22:39 -0400 (Wed, 09 May 2007) | 2 lines
Use the DTMF frame on the channel when returning a DTMF frame from AST_FRAME_NULL or AST_FRAME_VOICE.
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r63612 | russell | 2007-05-09 11:55:27 -0500 (Wed, 09 May 2007) | 5 lines
Modify ast_senddigit_begin() to use the same assumptions used elsewhere in the
code in that if a channel does not have a send_digit_begin() callback, it only
cares about DTMF END events. (pointed out by Michael Neuhauser on the
asterisk-dev list)
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r63608 | russell | 2007-05-09 11:43:50 -0500 (Wed, 09 May 2007) | 5 lines
Only call ast_senddigit_begin() in ast_senddigit() if the channel has a
send_digit_begin() callback. Checking the END_DTMF_ONLY flag was the
wrong thing to do, because that flag indicates that a *bridged* channel
only wants DTMF END events coming from this channel.
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r63286 | file | 2007-05-07 17:45:01 -0400 (Mon, 07 May 2007) | 10 lines
Merged revisions 63285 via svnmerge from
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r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2 lines
Properly handle what happens during a masquerade in relation to group counting. (issue #9657 reported by ramonpeek)
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r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) | 17 lines
Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending).
This set of changes came from a debugging session I had with Dwayne Hubbard.
When he called into his home FXO, ran the Echo application, and pressed a
digit, the digit would be echoed back and would never end. This is fixed,
along with a couple other little improvements.
* When chan_zap is in the middle of playing a digit to a channel, it feeds
back null frames, not voice frames. So, I have modified ast_read to check
the timing on emulated DTMF when it receives null frames, in addition to
where it was doing this on voice frames.
* Make a tweak to setting the duration on emulated DTMF digits. If there was
no duration specified, it set it to be the minimum, instead of the default.
* Instead of timing the emulated digits off of the number of samples in audio
frames that pass through, just use time values. Now there is no code in this
section that assumes 8kHz audio.
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r62789 | russell | 2007-05-02 17:59:09 -0500 (Wed, 02 May 2007) | 20 lines
Merge changes from team/russell/inband_dtmf ...
Fix some issues related to generating inband DTMF. There are two changes here:
1) The list of DTMF tones in the senddigit_begin() function explicitly
specified 100ms of the tone followed by 100ms of silence. This really
broke things with the way that Asterisk now wants complete control
over when the digit begins and ends. So, regardless of what Asterisk
really wanted to do, this was going to play out the tone at the length it
wanted to. This caused various problems like DTMF translation to inband to
be extremely unreliable.
The list of tones has been changed so that the correct DTMF tone is played
indefinitely until Asterisk tells it to stop.
2) ast_write() had to be modified to let a DTMF_END frame get processed even
when a generator is present. This is how the tone will finally get stopped.
(issues #8944, #9250, #9348, maybe others. Thanks to mdu113 from #8944 for
the testing and feedback!)
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r62689 | murf | 2007-05-02 11:10:50 -0600 (Wed, 02 May 2007) | 1 line
a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS.
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r62005 | file | 2007-04-25 22:19:51 -0500 (Wed, 25 Apr 2007) | 2 lines
Missed an ast_app_group_discard during merge. Thanks blitzrage!
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61805 | file | 2007-04-25 15:21:54 -0400 (Wed, 25 Apr 2007) | 10 lines
Merged revisions 61804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2 lines
Merge rewritten group counting support. No more storing data on the variable list of the channels. That was bad, mmmk? (issue #7497 reported by sabbathbh)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61806 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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