https://origsvn.digium.com/svn/asterisk/branches/1.4
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r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010) | 6 lines
When StopMonitor is called, ensure that it will not be restarted by a channel event.
(closes issue #16590)
Reported by: kkm
Patches:
resmonitor-16590-trunk.239289.diff uploaded by kkm (license 888)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258776 65c4cc65-6c06-0410-ace0-fbb531ad65f3
"Bad Things" would happen if Asterisk was compiled with DEBUG_THREADS, but a
loaded module was not (or vice versa). This also immensely simplifies the
lock code, since there are no longer 2 separate versions of them.
Review: https://reviewboard.asterisk.org/r/508/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added a new manager command to mute/unmute MixMonitor audio on a channel.
Added a new feature to audiohooks so that you can mute either read / write
(or both) types of frames - this allows for MixMonitor to mute either side
of the conversation without affecting the conversation itself.
(closes issue #16740)
Reported by: jmls
Review: https://reviewboard.asterisk.org/r/487/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some platforms prefix externally-visible symbols in object files generated
from C sources (most commonly, '_' is the prefix). On these platforms,
the existing symbol export filtering process ends up suppressing all the symbols
that are supposed to be left visible. This patch allows the prefix string
to be supplied to the top-level Makefile in the LINKER_SYMBOL_PREFIX variable,
and then generates the linker scripts as required to include the prefix
supplied.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar 2010) | 44 lines
Several fixes regarding RFC2833 DTMF detection.
Here is a copy and paste of the details from my request on
reviewboard that dealt with these changes:
Fix 1. The first change in place is to fix Mantis issue 15811, which deals with a situation where Asterisk will incorrectly interpret out of order RFC2833 frames as duplicate DTMF digits. For instance, we would receive a sequence like:
seqno 1: DTMF 1
seqno 2: DTMF 1
seqno 3: DTMF 1
seqno 4: DTMF 1
seqno 6: DTMF 1 (end)
seqno 5: DTMF 1
seqno 7: DTMF 1 (end)
seqno 8: DTMF 1 (end)
Prior to this patch when we received the frame with seqno 5, we would interpret this as a new DTMF 1. With this patch, we will check the seqno of the incoming digit and not process the frame if the seqno is lower than the last recorded seqno. Note that we do not record the seqno of the dropped DTMF frame for future processing. While the above situation is what was designed to be fixed, the patch is written in such a way that the following would also be fixed too:
seqno 9: DTMF 1
seqno 10: DTMF 1 (end)
seqno 11: DTMF 1 (end)
seqno 13: DTMF 2
seqno 12: DTMF 1 (end)
seqno 14: DTMF 2
seqno 15: DTMF 2 (end)
seqno 16: DTMF 2 (end)
seqno 17: DTMF 2 (end)
In this second situation, the beginning of the DTMF 2 arrives before the final end frame of the DTMF 1. With the patch, seqno 12 is no processed and thus we properly interpret the DTMF.
Fix 2. The second change in place is to fix an issue like the following:
seqno 1: DTMF 1
seqno 2: DTMF 1
seqno 3: DTMF 1 (end) *packet lost*
seqno 4: DTMF 1 (end) *packet lost*
seqno 5: DTMF 1 (end) *packet lost*
seqno 6: DTMF 2
When we receive seqno 6, we had code in place that was supposed to properly end the previously unended DTMF 1. The problem was that the code was essentially a no-op. The code would set up an end frame for the DTMF 1 but would immediately overwrite the frame with the begin for DTMF 2. I changed process_dtmf_rfc2833() so that instead of returning a single frame, it is given as an output parameter a list of frames. Each frame that needs to be returned is appended to this list.
Fix 3. The final change is a minor one where an AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco DTMF or an RFC 3389 frame and no frame was returned, then we would return &ast_null_frame. The problem is that earlier in the function, we may have generated an AST_CONTROL_SRCCHANGE frame and put it in the list of frames we wish to return. This frame would be lost in such a case. The patch fixes this problem
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
application is executing on a channel.
This patch addresses an issue found during working with end-users
using res_fax. If an incoming call is answered in the dialplan, or
jumps to the 'fax' extension due to reception of a CNG tone (with
faxdetect enabled), and then the remote endpoint sends a T.38
re-INVITE, it is possible for the channel's T.38 state to be
'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately,
even if the application wants to use T.38, it can't respond to the
peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS
control frame that chan_sip sent originally has been lost, and the
application needs the content of that frame to be able to formulate a
reply.
This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS,
AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip
will re-send the original control frame (with
AST_T38_REQUEST_NEGOTIATE as the request type), and the application
can respond as normal. If this occurs within the five second timeout
in chan_sip, the automatic cancellation of the peer reinvite will be
stopped, and the application will 'own' the negotiation process from
that point onwards.
This also improves the code path in chan_sip to allow sip_indicate(),
when called for AST_CONTROL_T38_PARAMETERS, to be able to return a
non-zero response, which should have been in place before since the
control frame *can* fail to be processed properly. It also modifies
ast_indicate() to return whatever result the channel driver returned
for this control frame, rather than converting all non-zero results
into '-1'. Finally, the new request type intentionally returns a
positive value, so that an application that sends
AST_T38_REQUEST_PARMS can know for certain whether the channel driver
accepted it and will be replying with a control frame of its own, or
whether it was ignored (if the sip_indicate()/ast_indicate() path had
properly supported failure responses before, this would not be
necessary).
This patch also modifies res_fax to take advantage of the new request.
In addition, this patch makes sip_t38_abort() actually lock the
private structure before doing its work... bad programmer, no donut.
This patch also enhances chan_sip's 'faxdetect' support to allow
triggering on T.38 re-INVITEs received as well as CNG tone detection.
Review: https://reviewboard.asterisk.org/r/556/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010) | 72 lines
Ensure that monitor recordings are written to the correct location (again)
This is an extension to 248860. As such the dialplan test has been extended:
; non absolute path, not combined
exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test)
exten => 5040, n, dial(sip/5001)
; absolute path, not combined
exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2)
exten => 5041, n, dial(sip/5001)
; no path, not combined
exten => 5042, 1, monitor(wav,monitor_test3)
exten => 5042, n, dial(sip/5001)
; combined: changemonitor from non absolute to no path (leaves tmp/jeff)
exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m)
exten => 5043, n, changemonitor(monitor_test5)
exten => 5043, n, dial(sip/5001)
; combined: changemonitor from no path to non absolute path
exten => 5044, 1, monitor(wav,monitor_test6,m)
exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this wasn't possible before
exten => 5044, n, dial(sip/5001)
; non absolute path, combined
exten => 5045, 1, monitor(wav,tmp/jeff/monitor_test8,m)
exten => 5045, n, dial(sip/5001)
; absolute path, combined
exten => 5046, 1, monitor(wav,/tmp/jeff/monitor_test9,m)
exten => 5046, n, dial(sip/5001)
; no path, combined
exten => 5047, 1, monitor(wav,monitor_test10,m)
exten => 5047, n, dial(sip/5001)
; combined: changemonitor from non absolute to absolute (leaves tmp/jeff)
exten => 5048, 1, monitor(wav,tmp/jeff/monitor_test11,m)
exten => 5048, n, changemonitor(/tmp/jeff/monitor_test12)
exten => 5048, n, dial(sip/5001)
; combined: changemonitor from absolute to non absolute (leaves /tmp/jeff)
exten => 5049, 1, monitor(wav,/tmp/jeff/monitor_test13,m)
exten => 5049, n, changemonitor(tmp/jeff/monitor_test14)
exten => 5049, n, dial(sip/5001)
; combined: changemonitor from no path to absolute
exten => 5050, 1, monitor(wav,monitor_test15,m)
exten => 5050, n, changemonitor(/tmp/jeff/monitor_test16)
exten => 5050, n, dial(sip/5001)
; combined: changemonitor from absolute to no path (leaves /tmp/jeff)
exten => 5051, 1, monitor(wav,/tmp/jeff/monitor_test17,m)
exten => 5051, n, changemonitor(monitor_test18)
exten => 5051, n, dial(sip/5001)
; not combined: changemonitor from non absolute to no path (leaves tmp/jeff)
exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19)
exten => 5052, n, changemonitor(monitor_test20)
exten => 5052, n, dial(sip/5001)
; not combined: changemonitor from no path to non absolute
exten => 5053, 1, monitor(wav,monitor_test21)
exten => 5053, n, changemonitor(tmp/jeff/monitor_test22)
exten => 5053, n, dial(sip/5001)
; not combined: changemonitor from non absolute to absolute (leaves tmp/jeff)
exten => 5054, 1, monitor(wav,tmp/jeff/monitor_test23)
exten => 5054, n, changemonitor(/tmp/jeff/monitor_test24)
exten => 5054, n, dial(sip/5001)
; not combined: changemonitor from absolute to non absolute (leaves /tmp/jeff)
exten => 5055, 1, monitor(wav,/tmp/jeff/monitor_test24)
exten => 5055, n, changemonitor(tmp/jeff/monitor_test25)
exten => 5055, n, dial(sip/5001)
; not combined: changemonitor from no path to absolute
exten => 5056, 1, monitor(wav,monitor_test26)
exten => 5056, n, changemonitor(/tmp/jeff/monitor_test27)
exten => 5056, n, dial(sip/5001)
; not combined: changemonitor from absolute to no path (leaves /tmp/jeff)
exten => 5057, 1, monitor(wav,/tmp/jeff/monitor_test28)
exten => 5057, n, changemonitor(monitor_test29)
exten => 5057, n, dial(sip/5001)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254277 65c4cc65-6c06-0410-ace0-fbb531ad65f3
users expect them to work.
'core set debug' and 'core set verbose' can optionally change the
level for a specific filename; however, this is actually for a
specific source file name, not the module that source file is included
in. With examples like chan_sip, chan_iax2, chan_misdn and others
consisting of multiple source files, this will not lead to the
behavior that users expect. If they want to set the debug level for
chan_sip, they want it set for all of chan_sip, and not to have to
also set it for reqresp_parser and other files that comprise the
chan_sip module.
This patch changes this functionality to be module-name based instead
of file-name based.
To make this work, some Makefile modifications were required to ensure
that the AST_MODULE definition is present in each object file produced
for each module as well.
Review: https://reviewboard.asterisk.org/r/574/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The other issue mentioned in this bug will be more difficult to resolve since we
have no idea (right now) of knowing if the command that is aliased has been
installed yet.
(issue #16978)
Reported by: jw-asterisk
Tested by: seanbright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously, values that began with whitespace were silently treated as 'no',
and all non-'yes' values were also treated as 'no'. Now the supplied value
is specifically checked for a 'yes' or 'no' (or equivalent) value, after skipping
leading whitespace. If the value is not valid, then a warning message is generated.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change basically reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times when we
detect that the other side of the conversation has changed the ssrc.
The problem is that SRCUPDATE control frames are sent many times where
we don't want a new ssrc, including whenever Asterisk has to send DTMF
in a normal bridge. This is also not the first time that this mistake
has been made. The initial implementation of the ast_rtp_new_source
function also changed the ssrc--and then it was removed because of
this same issue. Then, we put it back in again to fix a different
issue. This patch attempts to only change the ssrc when we see that
the other side of the conversation has changed the ssrc.
It also renames some functions to make their purpose more clear.
Review: https://reviewboard.asterisk.org/r/540/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010) | 18 lines
Ensure that monitor recordings are written to the correct location (again)
This is an extension to 248757. As such the dialplan test has been extended:
exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b)
exten => 5040, n, dial(sip/5001)
exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b)
exten => 5041, n, dial(sip/5001)
exten => 5042, 1, monitor(wav,monitor_test3,b)
exten => 5042, n, dial(sip/5001)
exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test3,m)
exten => 5043, n, changemonitor(monitor_test4)
exten => 5043, n, dial(sip/5001)
exten => 5044, 1, monitor(wav,monitor_test4,m)
exten => 5044, n, changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by design and emits a warning
exten => 5044, n, dial(sip/5001)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010) | 15 lines
Ensure that monitor recordings are written to the correct location.
Recordings should be placed in the monitor directory when a non-absolute path
is used.
Exact dialplan used for testing:
exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b)
exten => 5040, n, dial(sip/5001)
exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b)
exten => 5041, n, dial(sip/5001)
exten => 5042, 1, monitor(wav,monitor_test3,b)
exten => 5042, n, dial(sip/5001)
ABE-2101
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The new JabberStatus event gives a concise view of the status change to the AMI
clients. Thanks fiddur!
(closes issue #16760)
Reported by: fiddur
Patches:
244498.2.diff uploaded by fiddur (license 678)
Tested by: fiddur, phsultan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Detect all platforms that don't like that, either, and ensure that when documentation is
missing, we pass a non-NULL pointer when outputting the corresponding documentation.
(closes issue #16689)
Reported by: bklang
Patches:
20100209__issue16689__with_tests.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/497/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Initialize the calendars container before calling load_config and return FAILURE
on allocation failure. Also, use the AST_MODULE_LOAD_* values for return values.
Thanks to rmudgett for pointing out the error and the need to use the defined
values for return
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r242520 | tilghman | 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010) | 8 lines
Only rebuild bison and flex source files on demand, if bison and flex are detected by the configure script.
Changed after discussion on the -dev list about possible unnecessary build
failures, due to checkouts/untars causing these special source files to
possibly be newer than their resulting C files. This should additionally
ensure that nobody need learn about extra Makefile arguments to ensure the
proper files get rebuilt when changes are made to these special source files.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242521 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010) | 16 lines
Add a flag to disable the Background behavior, for AGI users.
This is in a section of code that relates to two other issues, namely
issue #14011 and issue #14940), one of which was the behavior of
Background when called with a context argument that matched the current
context. This fix broke FreePBX, however, in a post-Dial situation.
Needless to say, this is an extremely difficult collision of several
different issues. While the use of an exception flag is ugly, fixing all
of the issues linked is rather difficult (although if someone would like
to propose a better solution, we're happy to entertain that suggestion).
(closes issue #16434)
Reported by: rickead2000
Patches:
20091217__issue16434.diff.txt uploaded by tilghman (license 14)
20091222__issue16434__1.6.1.diff.txt uploaded by tilghman (license 14)
Tested by: rickead2000
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237406 65c4cc65-6c06-0410-ace0-fbb531ad65f3