mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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1.6.0.8
1.6.0.9
1.6.1-beta1
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1.6.1-beta3
1.6.1-beta4
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10.0.0
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10.1.1
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10.10.0
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10.11.0
10.11.0-digiumphones
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10.11.1-digiumphones
10.12.0
10.12.0-digiumphones
10.12.0-digiumphones-rc1
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10.12.0-rc1
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10.12.1
10.12.1-digiumphones
10.12.2
10.12.2-digiumphones
10.12.3
10.12.3-digiumphones
10.12.4
10.12.4-digiumphones
10.2.0
10.2.0-rc1
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10.2.1
10.3.0
10.3.0-rc1
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10.3.1
10.4.0
10.4.0-digiumphones-rc1
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10.5.0
10.5.0-digiumphones
10.5.0-digiumphones-rc1
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10.5.0-rc1
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10.5.1
10.5.1-digiumphones
10.5.2
10.5.2-digiumphones
10.6.0
10.6.0-digiumphones
10.6.0-digiumphones-rc1
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10.6.0-rc1
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10.6.1
10.6.1-digiumphones
10.7.0
10.7.0-digiumphones
10.7.0-digiumphones-rc1
10.7.0-rc1
10.7.1
10.7.1-digiumphones
10.8.0
10.8.0-digiumphones
10.8.0-digiumphones-rc1
10.8.0-digiumphones-rc2
10.8.0-rc1
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10.9.0
10.9.0-digiumphones
10.9.0-digiumphones-rc1
10.9.0-digiumphones-rc2
10.9.0-digiumphones-rc3
10.9.0-rc1
10.9.0-rc2
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11.0.0
11.0.0-beta1
11.0.0-beta2
11.0.0-rc1
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11.0.1
11.0.2
11.1.0
11.1.0-rc1
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11.1.0-rc3
11.1.1
11.1.2
11.10.0
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11.10.2
11.11.0
11.11.0-rc1
11.12.0
11.12.0-rc1
11.12.1
11.13.0
11.13.0-rc1
11.13.1
11.14.0
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11.14.1
11.14.2
11.15.0
11.15.0-rc1
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11.16.0
11.16.0-rc1
11.17.0
11.17.0-rc1
11.17.1
11.18.0-rc1
11.19.0-rc1
11.2.0
11.2.0-rc1
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11.2.1
11.2.2
11.20.0-rc1
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11.20.0-rc3
11.21.0
11.21.0-rc1
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11.21.1
11.22.0-rc1
11.3.0
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11.4.0
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11.5.0
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11.5.1
11.6-cert11
11.6.0
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11.6.1
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11.8.0
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11.8.1
11.9.0
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12.0.0
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12.1.0
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12.1.1
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12.3.0
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12.3.1
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12.4.0
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12.5.0
12.5.0-rc1
12.5.1
12.6.0
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12.6.1
12.7.0
12.7.0-rc1
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12.7.1
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12.8.0
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12.8.1
12.8.2
13.0.0
13.0.0-beta1
13.0.0-beta2
13.0.0-beta3
13.0.1
13.0.2
13.1-cert2
13.1.0
13.1.0-rc1
13.1.0-rc2
13.1.1
13.2.0
13.2.0-rc1
13.2.1
13.3.0
13.3.0-rc1
13.3.1
13.3.2
13.4.0-rc1
13.5.0-rc1
13.6.0-rc1
13.6.0-rc2
13.6.0-rc3
13.7.0
13.7.0-rc1
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13.7.0-rc3
13.7.1
13.8.0-rc1
certified/1.8.11-cert1
certified/1.8.11-cert10
certified/1.8.11-cert2
certified/1.8.11-cert3-rc1
certified/1.8.11-cert3-rc2
certified/1.8.11-cert4
certified/1.8.11-cert5
certified/1.8.11-cert5-rc1
certified/1.8.11-cert5-rc2
certified/1.8.11-cert6
certified/1.8.11-cert7
certified/1.8.11-cert8
certified/1.8.11-cert9
certified/1.8.11-cert9-rc1
certified/1.8.15-cert1
certified/1.8.15-cert1-rc1
certified/1.8.15-cert1-rc2
certified/1.8.15-cert1-rc3
certified/1.8.15-cert2
certified/1.8.15-cert3
certified/1.8.15-cert4
certified/1.8.15-cert5
certified/1.8.15-cert6
certified/1.8.15-cert7
certified/1.8.28-cert1
certified/1.8.28-cert1-rc1
certified/1.8.28-cert2
certified/1.8.28-cert3
certified/1.8.28-cert4
certified/1.8.28-cert5
certified/1.8.6-cert1
certified/11.2-cert1
certified/11.2-cert1-rc1
certified/11.2-cert1-rc2
certified/11.2-cert2
certified/11.2-cert3
certified/11.6-cert1
certified/11.6-cert1-rc1
certified/11.6-cert1-rc2
certified/11.6-cert10
certified/11.6-cert11
certified/11.6-cert12
certified/11.6-cert2
certified/11.6-cert3
certified/11.6-cert4
certified/11.6-cert5
certified/11.6-cert6
certified/11.6-cert7
certified/11.6-cert8
certified/11.6-cert9
certified/13.1-cert1
certified/13.1-cert1-rc1
certified/13.1-cert1-rc2
certified/13.1-cert1-rc3
certified/13.1-cert2
certified/13.1-cert3
${ noResults }
1017 Commits (21ca773c2829d7448a8a5dfec5f81336714ee590)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
04836d554d |
Fix handling of backreferences for ENUM lookups
enum.c did not handle regex backtraces correctly. The '\1' in the regex is a backreference that requires a pattern match to be inserted. The way the code used to work is that it would find the backreference and insert the entire input string minus the '+'. This is incorrect. The regexec() function takes in a variable called pmatch which is an array of structs containing the start and end indexes for each backreference substring. The original code actually passed the pmatch array pointer into regexec but never did anything with it. Now when a backtrace is found, the backtrace number is looked up in the pmatch array and the correct substring is inserted. (closes issue #14576) Reported by: chris-mac Review: http://reviewboard.digium.com/r/187/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@180532 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
5436d8709f |
Fix problems when RTP packet frame size is changed
During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@180372 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
c42b21bc6a |
Look for the number in a callerid string starting from the end. This way a value using <> can exist in the name portion.
(issue #AST-194) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@180194 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
604a51f341 |
These changes allow AEL to better check ${} constructs within $[...], that are concatenated with text.
I modified and added rules in ast_expr2.fl to better handle the concatenations. I added some default routines to ast_expr2.y so the standalone would compile. It also looks like I haven't run this thru bison since 2.1, so it's good to get this updated. The Makefile has comments added now for check_expr2 and check_expr to explain what they are for, and how to run them. The testexpr2s stuff has been removed, in favor of check_expr2. expr2.testinput has been updated to include the two expressions that inspired these changes (from mcnobody on #asterisk this morning) The regression has been run and all looks well. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@179807 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
2e4471d758 |
Ensure chan->fdno always gets reset to -1 after handling a channel fd event.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@179741 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
3f2a1247f4 |
Move where fdno is set to the default value to *after* the read callback of 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@179671 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
a1d249577e |
Make it easier to detect an improper call to ast_read().
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@179608 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
4055ec6c57 |
Fix bridging regression from commit 176701
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@179536 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
b705454875 |
When ending a recording with silence detection, remember to reduce the duration.
The end of the recording is correspondingly trimmed, but the duration was not trimmed by the number of seconds trimmed, so the saved duration was necessarily longer than the actual soundfile duration. (closes issue #14406) Reported by: sasargen Patches: 20090226__bug14406.diff.txt uploaded by tilghman (license 14) Tested by: sasargen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@179468 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
6706e0be24 |
Ensure that only one thread is calling ast_settimeout() on a channel at a time.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@179461 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
5062dff9dd |
Remove several silly warnings in editline. One about a broken preprocessor directive, and another about strlcpy/strlcat.
(closes issue #14264) Reported by: dimas git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@179395 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
|
c4d2dc7967 |
This patch completes the fixes nec. to make 1.4 asterisk dialplan expressions ($[...]) 8-bit transparent
While I was updating ast_expr2.fl, I missed one rule that would allow 8-bit chars to be caught
in tokens; and in so doing, it absorbs the ${ sequence and messes up the
checking of raw exprs by AEL.
Trunk already has these changes.
(closes issue #14543)
Reported by: klaus3000
Patches:
patch.14543 uploaded by murf (license 17)
Tested by: murf
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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18 years ago |
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7374b9195f |
Update the copyright year for the main page of the doxygen documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178508 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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da0d84c3e4 |
Only set dtmfcount on BEGIN, and ensure it gets reset to 0 properly.
(issue #14460) Reported by: moliveras Tested by: russell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178373 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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aa5a927b69 |
Fix infinite DTMF when a BEGIN is received without an END.
This commit is related to rev 175124 of 1.4 where a previous attempt was made to fix this problem. The problem with the previous patch was that the inserted code needed to go _before_ setting the lastrxts to the current timestamp. Because those were the same, the dtmfcount variable was never decremented, and so the END was never sent. In passing, I removed the dtmfsamples variable which was completed unused. I also removed a redundant setting of the lastrxts variable. (closes issue #14460) Reported by: moliveras git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178141 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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4af0175285 |
Don't print the CR-NL combination when we aren't outputting to the manager.
An embedded CR-NL in a CLI command screws up several AMI parsers that don't expect to see that combination in the middle of output. (Closes issue #14305) Reported by: martins Patch by: tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177786 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d290eed8e0 |
This patch fixes a problem with 8-bit input to the ast_expr2 scanner.
The real culprit was the --full argument to flex in the Makefile! This causes a 7-bit scanner to be generated. I reviewed the rules and found one rule where I needed to specifically include 8-bit chars for a token. I tested against the text supplied by ibercom, and all looks very well. This has been there a surprisingly long time! (closes issue #14498) Reported by: ibercom Patches: 14498.patch uploaded by murf (license 17) Tested by: murf git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177540 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a591301b89 |
Merged revisions 177035 manually from
https://origsvn.digium.com/svn/asterisk/trunk ........ r177035 | dbailey | 2009-02-18 11:24:07 -0600 (Wed, 18 Feb 2009) | 2 lines Fixed error where a check for an zero length, terminated string was needed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@177039 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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bf0b8526b1 |
Need to take into account the \0 terminator of the old string to determine the amount available.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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de8f6bab86 |
Modify bridging to properly evaluate DTMF after first warning is played
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@176701 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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4d808232cf |
correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176254 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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1b5b3efcae |
fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176216 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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db4dc67740 |
fix mis-spelling of the word registered.
Reported by De_Mon on #asterisk-dev. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175921 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7accd1ec46 |
Fix a place where filestreams were not refcounted properly
This section was already present in trunk and other branches, but did not exist in 1.4. (closes issue #14395) Reported by: ZX81 Patches: 14395.patch uploaded by putnopvut (license 60) Tested by: ZX81 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175407 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a13deff994 |
Fix crashes when receiving certain T.38 packets. Also, increase the maximum
size of T.38 packets and warn users when they try to set the limits above those maximums. (closes issue #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt uploaded by Corydon76 (license 14) Tested by: schern git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175311 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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bfaa341f58 |
Don't send DTMF for infinite time if we do not receive an END event.
I thought that this was going to end up being a pretty gnarly fix, but it turns
out that there was actually already a configuration option in rtp.conf,
dtmftimeout, that was intended to handle this situation. However, in between
Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost.
So, this commit brings it back to life.
The default timeout is 3 seconds. However, it is worth noting that having
this be configurable at all is not really the recommended behavior in RFC 2833.
From Section 3.5 of RFC 2833:
Limiting the time period of extending the tone is necessary
to avoid that a tone "gets stuck". Regardless of the
algorithm used, the tone SHOULD NOT be extended by more than
three packet interarrival times. A slight extension of tone
durations and shortening of pauses is generally harmless.
Three seconds will pretty much _always_ be far more than three packet
interarrival times. However, that behavior is not required, so I'm going to
leave it with our legacy behavior for now.
Code from svn/asterisk/team/russell/issue_14460
(closes issue #14460)
Reported by: moliveras
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@175124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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18 years ago |
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70f7c7e9cb |
Revert RTP changes for continuation of DTMF. Proxy commit by russell via SMS.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174997 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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1d4e4ff3d1 |
Clear out the current event after forcing the end of a digit
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174986 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9dff8995b4 |
Fixify infinite DTMF in the case that no RFC2833 END event is ever received
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174985 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9e38dd5427 |
Restore a behavior that was recently changed, when we fixed issue #13962 and
issue #13363 (related to issue #6176). When a hangup occurs during a Macro execution in earlier 1.4, the h extension would execute within the Macro context, whereas it was always supposed to execute only within the main context (where Macro was called). So this fix checks for an "h" extension in the deepest macro context where a hangup occurred; if it exists, that "h" extension executes, otherwise the main context "h" is executed. (closes issue #14122) Reported by: wetwired Patches: 20090210__bug14122.diff.txt uploaded by Corydon76 (license 14) Tested by: andrew git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174885 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b60ec2baa1 |
Improve behavior of jitterbuffer when maxjitterbuffer is set.
This change improves the way the jitterbuffer handles maxjitterbuffer and dramatically reduces the number of frames dropped when maxjitterbuffer is exceeded. In the previous jitterbuffer, when maxjitterbuffer was exceeded, all new frames were dropped until the jitterbuffer is empty. This change modifies the code to only drop frames until maxjitterbuffer is no longer exceeded. Also, previously when maxjitterbuffer was exceeded, dropped frames were not tracked causing stats for dropped frames to be incorrect, this change also addresses that problem. (closes issue #14044) Patches: bug14044-1.diff uploaded by mnicholson (license 96) Tested by: mnicholson Review: http://reviewboard.digium.com/r/144/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@174583 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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4e069885ce |
Fix feature inheritance with builtin features
When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172517 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c257ffeed0 |
Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if
Asterisk runs as a non-root user and the administrator does a 'restart now', Asterisk loses the ability to set QOS on packets. (closes issue #14004) Reported by: nemo Patches: 20090105__bug14004.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172438 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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13a60eba0c |
This patch fixes h-exten running misbehavior in manager-redirected
situations. What it does: 1. A new Flag value is defined in include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the bridge hangup exten code not to run the h-exten there (nor publish the bridge cdr there). It will done at the pbx-loop level instead. 2. In the manager Redirect code, I set this flag on the channel if the channel has a non-null pbx pointer. I did the same for the second (chan2) channel, which gets run if name2 is set... and the first succeeds. 3. I restored the ending of the cdr for the pbx loop h-exten running code. Don't know why it was removed in the first place. 4. The first attempt at the fix for this bug was to place code directly in the async_goto routine, which was called from a large number of places, and could affect a large number of cases, so I tested that fix against a fair number of transfer scenarios, both with and without the patch. In the process, I saw that putting the fix in async_goto seemed not to affect any of the blind or attended scenarios, but still, I was was highly concerned that some other scenarios I had not tested might be negatively impacted, so I refined the patch to its current scope, and jmls tested both. In the process, tho, I saw that blind xfers in one situation, when the one-touch blind-xfer feature is used by the peer, we got strange h-exten behavior. So, I inserted code to swap CDRs and to set the HANGUP_DONT field, to get uniform behavior. 5. I added code to the bridge to obey the HANGUP_DONT flag, skipping both publishing the bridge CDR, and running the h-exten; they will be done at the pbx-loop (higher) level instead. 6. I removed all the debug logs from the patch before committing. 7. I moved the AUTOLOOP set/reset in the h-exten code in res_features so it's only done if the h-exten is going to be run. A very minor performance improvement, but technically correct. (closes issue #14241) Reported by: jmls Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17) Tested by: murf, jmls git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172030 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0b74f727d7 |
Prevent a crash from occurring when a jitter buffer interpolated frame is
removed from a slinfactory slinfactory used the "samples" field of an ast_frame in order to determine the amount of data contained within the frame. In certain cases, such as jitter buffer interpolated frames, the frame would have a non-zero value for "samples" but have NULL "data" This caused a problem when a memcpy call in ast_slinfactory_read would attempt to access invalid memory. The solution in use here is to never feed frames into the slinfactory if they have NULL "data" (closes issue #13116) Reported by: aragon Patches: 13116.diff uploaded by putnopvut (license 60) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@171621 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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87c02936b6 |
When a channel is answered make sure any indications currently playing stop. 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@170648 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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1bef118f00 |
Fix broken call pickup
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@170392 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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3ff70ed000 |
Don't crash if RTCP is not enabled on an RTP structure but statistics are output.
(closes issue #14234) Reported by: jcovert Patches: rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551) rtp.c.patch-svn-165599 uploaded by jcovert (license 551) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@170239 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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4ee4e941f8 |
Do a string comparison instead of pointer comparison since some people specify the context they are actually in as an argument to get around some funkiness.
(closes issue #14011) Reported by: dveiga Patches: pbx.c.patch uploaded by dveiga (license 665) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@170050 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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376d85f96c |
Read lock the contexts to maintain the locking order when we are notified that the state of a device has changed.
(closes issue #13839) Reported by: mcallist git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@169867 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9db9bae4d9 |
Adding revision 169794 to 1.4 since 1.4 is also affected by the issue
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@169797 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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e34da1e519 |
Extra NULLs in the output cause some terminal types to abort in the middle of
a color code, causing terminal weirdness. (closes issue #14130) Reported by: coolmig Patches: 20090121__bug14130.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, coolmig git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@169722 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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f311539733 |
Truncate userevents at the end of a line, when the command exceeds the buffer.
(closes issue #14278) Reported by: fnordian git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@169364 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d671bb1404 |
Fix the conjugation of Russian and Ukrainian languages.
(related to issue #12475) Reported by: chappell Patches: vm_multilang.patch uploaded by chappell (license 8) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168828 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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99313c7b92 |
Don't read into a buffer without first checking if a value is beyond the end.
(closes issue #13600) Reported by: atis Patches: 20090106__bug13600.diff.txt uploaded by Corydon76 (license 14) Tested by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168603 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9161b7fc87 |
Revert unnecessary indications API change from rev 122314
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168561 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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5fe8bde41a |
Fix the last couple of places where free() was improperly used directly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@167566 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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2839e074ef |
Don't fclose() the file early, the filestream destructor will handle it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@167554 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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301945890d |
Only try to close the file if one was actually opened
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@167545 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c67d152525 |
Don't use free() directly. This caused a crash since ast_filestream is now an ao2 object.
Reported by JunK-Y on IRC, #asterisk-dev git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@167541 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |