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10.12.0
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10.12.1-digiumphones
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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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10.6.0
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10.7.0-digiumphones
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10.8.0
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11.2.1
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12.0.0
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13.0.0
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13.1-cert2
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13.1.1
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13.7.0
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13.7.1
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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 }
315 Commits (1.4)
Author | SHA1 | Message | Date |
---|---|---|---|
|
82506ffa01 |
The meetme CLI command completion leaves conferences mutex locked.
When issuing a meetme kick CLI command and an invalid (non-existent) conference number is specified, pressing Tab leaves the conferences mutex locked and, therefore, all conferences deadlock. Add missing unlock. (closes issue #19336) Reported by: zvision Patches: app_meetme.diff uploaded by zvision (license 798) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@320235 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
1fb86f03c8 |
Backport MeetMe related reference leaks fixes from 1.6.2/1.8/trunk.
I had forgotten that MeetMe in 1.4 also used astobj2, so backport the fixes where appropriate. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@304820 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
58a666ac5a |
Fix channel redirect out of MeetMe() and other issues with channel softhangup.
Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped working properly. This issue includes a patch that resolves the issue by removing a call to ast_check_hangup() from app_meetme.c. I left that in my patch, as it doesn't need to be there. However, the rest of the patch fixes this problem with or without the change to app_meetme. The key difference between what happens before and after this patch is the effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(), ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme sees this which causes it to exit as intended. Checking ast_check_hangup() caused app_meetme to exit earlier in the process, and the target of the redirect saw the condition where ast_read() returned NULL. Removing ast_check_hangup() works around the issue in app_meetme, but doesn't solve the issue if another application did the same thing. There are also other edge cases where if an application finishes at the same time that a redirect happens, the target of the redirect will think that the channel hung up. So, I made some changes in pbx.c to resolve it at a deeper level. There are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to abort the hangup process. My patch extends this to remove the END_OF_Q frame from the channel's read queue, making the "abort hangup" more complete. This same technique was used in every place where a softhangup flag was cleared. (closes issue #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: https://reviewboard.asterisk.org/r/1082/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@303546 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
92ea2179e5 |
Add "DAHDI" to a couple of app_meetme error messages.
This is in response to some questions on IRC. To the user, there was nothing that made it obvious that this error had anything to do with DAHDI not being loaded. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@297228 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
3702c0cc02 |
Fix misvalidation of meetme pins in conjunction with the 'a' MeetMe flag.
When using the 'a' MeetMe flag and having a user and admin pin setup for your conference, using the user pin would gain you admin priviledges. Also, when no user pin was set, an admin pin was, the 'a' MeetMe flag wasn't used, and the user tried to enter a conference then they were still prompted for a pin and forced to hit #. (closes issue #17908) Reported by: kuj Patches: pins_2.patch uploaded by kuj (license 1111) Tested by: kuj Review: [full review board URL with trailing slash] git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@287758 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
cf0d36226d |
Fix a dsp structure leak occuring when a local channel is put into a meetme
conference, then masquaraded away. ABE-2422 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@280341 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
c66d95c8e5 |
Make user removals and traversals thread safe in meetme.
Race conditions present in meetme involving the user list where a lack of locking has the potential for a user to be removed during a traversal or as in the case of the reporter after checking if the list is empty could cause a crash. Fixing this was done by convering the userlist to an ao2 container. (closes issue #17390) Reported by: Vince Review: https://reviewboard.asterisk.org/r/746/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@275773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
9bae102b7c |
Allow admin user to join conference without using admin mode and no user pin.
Configuring the conference in meetme.conf like the following: conf => 2345,,6666 did not prompt for pin when used without admin mode. This meant that the conference could not be joined as an admin even if the user knew the correct pin. The original bug report was submitted claiming that the blank user pin should deny entry into the conference. I think a better way to handle this would be with a feature enhancement that used the following syntax: conf => 2345,X,6666 - where X denotes no acceptable pin allowed (closes issue #15704) Reported by: modelnine git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273474 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
f5310e3bc7 |
Ensure channel placed in meetme in ringing state is properly hung up.
An outgoing channel placed in meetme while still ringing which was then hung up would not exit meetme and the channel was not properly destroyed. Specifically checking for this scenario by looking at the appropriate control frames resolves the issue. (closes issue #15871) Reported by: Ivan Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan (license 229) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273354 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
2f6313a1c8 |
First caller into a dynamic conference now enter pin once.
If MeetMe is configured to use dynamic conference numbers, then the first caller (which creates the conference) had to enter the PIN number twice. (closes issue #15878) Reported by: shawkris Patches: issue15878.patch uploaded by pabelanger (license 224) Tested by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@272255 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
dbcb9d1577 |
fixes app_meetme dsp error
We attempted to detect silence after translating a frame from signed linear. This caused a flooding of errors. To resolve this the code to detect silence was moved before the translation. (closes issue #17133) Reported by: jsdyer git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@262662 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
ef93e3343e |
Don't log to debug unless debug is turned on
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@248268 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
3d20d25e4c |
Avoid a crash with large numbers of MeetMe conferences.
Similar to changes made to Queue(), when we have large numbers of conferences in meetme.conf (1000s) and we use alloca()/strdupa(), we can blow out the stack and crash, so instead just use a single fixed buffer. (closes issue #16509) Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded by seanbright (license 71) Tested by: seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@236509 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
d97debdbf5 |
Fix talking detection status after conference user is muted.
This patch ensures that when a conference user is muted that the accompanying AMI Meetme talking off event is sent. Also, the meetme list output is updated to show the muted user as unmonitored. (closes issue #16247) Reported by: dimas Patches: v3-16247.patch uploaded by dimas (license 88) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@234379 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
6e8a455534 |
Fix documentation for ast_softhangup() and correct the misuse thereof.
(closes issue #16103) Reported by: majorbloodnok git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@225105 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b10343fd19 |
When MOH is playing on the channel, announcements sent through the conference are not heard.
(closes issue #14588) Reported by: voipas Patches: 20090716__issue14588__2.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, twisted, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@217156 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
63cc189747 |
AST-2009-005
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@211528 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
94fa4d11b5 |
Improve support for media paths that can generate multiple frames at once.
There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@200991 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
65494bfdf7 |
Fix a bug where the MeetMe option 'D' did not actually prompt for the pin.
(closes issue #15050) Reported by: pmhaddad git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@195635 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
59f867a5cb |
revert commit that included extranous changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182807 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
f1f417a9d8 |
Improve the build system to *properly* remove unnecessary symbols from the runtime global namespace. Along the way, change the prefixes on some internal-only API calls to use a common prefix.
With these changes, for a module to export symbols into the global namespace, it must have *both* the AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows the linker to leave the symbols exposed in the module's .so file (see res_odbc.exports for an example). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182802 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
21ca773c28 |
Fix malloc debug macros to work properly with h323.
The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. Because using the ast prefix calls are a better choice, ast_free_ptr is the new wrapper for free to pass to functions. Also, a little bit of clean up was done to avoid the debug macros intentionally being redefined. (closes issue #13593) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181133 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
dadbbb0a56 |
Move ast_waitfor() down to avoid the results of the API call becoming stale.
This call to ast_waitfor() was being done way too soon in this section of code. Specifically, there was code in between the call to waitfor and the code that uses the result that puts the channel in autoservice. By putting the channel in autoservice, the previous results of ast_waitfor() become meaningless, as the autoservice thread will do it's own ast_waitfor() and ast_read() on the channel. So, when we came back out of autoservice and eventually hit the block of code that calls ast_read() on the channel, there may not actually be any input on the channel available. Even though the previous call to ast_waitfor() in app_meetme said there was input, the autoservice thread has since serviced the channel for some period of time. This bug manifested itself while dvossel was doing some testing of MeetMe in Asterisk trunk. He was using the timerfd timing module. When the code hit ast_read() erroneously, it determined that it must have been called because of input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was the cause of the last legitimate call to ast_read() done by autoservice. In this test, an IAX2 channel was calling into the MeetMe conference. It was _much_ more likely to be seen with an IAX2 channel because of the way audio is handled. Every audio frame that comes in results in a call to ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify the channel thread that a frame is waiting to be handled. So, the chances of ast_waitfor() indicating that a channel needs servicing due to a timer event on an IAX2 event is very high. Finally, it is interesting to note that if a different timing interface was being used, this bug would probably not be noticed. When ast_read() is called and erroneously thinks that there is a timer event to handle, it calls the ast_timer_ack() function. The pthread and dahdi timing modules handle the ack() function being called when there is no event by simply ignoring it. In the case of the timerfd module, it results in a read() on the timer fd that will block forever, as there is no data to read. This caused Asterisk to lock up very quickly. Thanks to dvossel and mmichelson for the fun debugging session. :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@179532 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b30800adfb |
Remove unused variable and make dev-mode compilation happy
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176252 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5ae664d5aa |
Open the DAHDI pseudo device and set it to be nonblocking atomically
Apparently on FreeBSD, attempting to set the O_NONBLOCKING flag separately from opening the file was causing an "inappropriate ioctl for device" error. While I cannot fathom why this would be happening, I certainly am not opposed to making the code a bit more compact/efficient if it also fixes a bug. (closes issue #14482) Reported by: ys Patches: meetme.patch uploaded by ys (license 281) Tested by: ys git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@176249 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c8f9023696 |
If we are unable to request a DAHDI pseudo channel and we are using the user introduction without review option make sure it gets unset so other code does not blindly assume a DAHDI pseudo channel exists.
(closes issue #14282) Reported by: cheesegrits git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@170147 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
e0bec5d67d |
(closes issue #13480)
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@165991 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
a391fd9b1c |
Fix some memory leaks found while looking at how realtime
configs are handled. Also cleaned up some coding guidelines violations in app_realtime.c, mostly related to spacing git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@165255 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
5d985ce022 |
Fix an issue where callers on an incoming call on an SLA trunk would not hear ringback.
We need to make sure that we don't start writing audio to the trunk channel until we're actually ready to answer it. Otherwise, the channel driver will treat it as inband progress, even though all they are getting is silence. (closes issue #12471) Reported by: mthomasslo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@162286 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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161a05a3d1 |
(closes issue #13899)
Reported by: akkornel This fix is the result of a bug fix in ast_app_separate_args r124395. If an argument does not exist it should always be set to a null string rather than a null pointer. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@157365 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7e353eeaa1 |
If the SLA thread is not started, then reload causes a memory leak.
(closes issue #13889) Reported by: eliel Patches: app_meetme.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@156294 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
738ec1fb71 |
For whatever reason, gcc only warned me about the possible use of an uninitialized variable when compiling 1.6.1.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@156289 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c9cec47bf0 |
(closes issue #13173)
Reported by: pep This change adds an announce_thread responsible for playing announcements to an existing conference. This allows all announcing to be immediately stopped if necessary but more importantly allows other threads that need to play something to not block. There are multiple benefits to this, but the actual bug is for solving the scenario for a channel to be unusable after hang up for the entire duration of the parting announcement. The parting announcement can be extremely long depending on what the user recorded upon joining the conference. Reviewed by Russell on Review Board: http://reviewboard.digium.com/r/25/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@156178 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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2ebf0eee36 |
remove some more chan_zap references
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@137529 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a57dd05769 |
convert this module to be able to handle DAHDI or Zaptel (reported on asterisk-users, don't know how this got missed before)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@137188 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
7b61edde36 |
build against the now-typedef-free dahdi/user.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@134254 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
f9818af8dd |
Adds DAHDI support alongside Zaptel. DAHDI usage favored, but all Zap stuff should continue working. Release announcement to follow.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@122314 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
3ca503066a |
Flipflop the sections for two options, since the section for 'X' (exit context)
may otherwise absorb keypresses meant for 's' (admin/user menu). (closes issue #12836) Reported by: blitzrage Patches: 20080611__bug12836.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@122137 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c99280a59e |
Include the proper headers for using mkdir on FreeBSD.
(closes issue #12430) Reported by: ys Patches: app_meetme.c.diff uploaded by ys (license 281) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114167 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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fb51aa974e |
Create the directory where name recordings will go if it does not exist.
(closes issue #12311) Reported by: rkeene Patches: 12311-mkdir.diff uploaded by qwell (license 4) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114029 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
f3274be612 |
Fix a bug where when calls on the trunk side hang up while on hold, the state
is not properly reflected. (closes issue #11990, reported by anakaoka, patched by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@110163 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
d9e83f85a6 |
Add an additional check for setting conference parameter when using the marked user options. It was possible for it to return to a no listen/no talk state if a masquerade happened.
(closes issue #12136) Reported by: aragon git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@107637 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
a04b584272 |
Only start the SLA thread if SLA has actually been configured.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106895 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
0b110bb5e8 |
If entering a conference with the 'w' option ensure that we can't listen or speak until the marked user appears.
(closes issue #11835) Reported by: alanmcmillan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@103324 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
fa640604de |
Replace current spy architecture with backport of audiohooks. This should take care of current known spy issues.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@98972 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7a3ee5a7fc |
Don't do conferencing totally in Zaptel if Monitor is running on the channel.
(closes issue #11709) Reported by: BigJimmy Patches: patch-meetmerec uploaded by BigJimmy (license 371) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@97450 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
4a869fa0c3 |
Don't increment user count until after name has been recorded (if enabled).
Issue 11048, tested by pep. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@92617 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8315785663 |
Make sure we always close the conference fd if we have an open one.
Issue 11383, reported by markmhy, patch by eliel. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@90696 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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1884c603d6 |
When unloading app_meetme destroy any auto created contexts created by SLA.
(closes issue #11367) Reported by: eliel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89571 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
bd5b8a4014 |
Update the SLAStation application to account for the case where the SLA thread
has a call out to the station, but the user has pressed a line button to answer the call instead of picking up the handset. If they do, the phone sends out a new INVITE. So, the SLAStation app must check to see if it is picking up a ringing trunk, and ensure that the other stations stop ringing. (reported internally, patched by me, tested by mogorman) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89296 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |