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
1.6.1-beta2
1.6.1-beta3
1.6.1-beta4
1.6.1-rc1
1.6.1.0
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1.6.1.0-rc3
1.6.1.0-rc4
1.6.1.0-rc5
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1.6.1.10-rc1
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1.6.1.11
1.6.1.12
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1.6.1.13-rc1
1.6.1.14
1.6.1.15-rc1
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1.6.1.16
1.6.1.17
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1.8.15-cert4
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1.8.9.0-rc3
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10.10.0
10.10.0-digiumphones
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10.12.2-digiumphones
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10.12.3-digiumphones
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10.2.0
10.2.0-rc1
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10.4.0-digiumphones-rc1
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10.5.2-digiumphones
10.6.0
10.6.0-digiumphones
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10.6.1
10.6.1-digiumphones
10.7.0
10.7.0-digiumphones
10.7.0-digiumphones-rc1
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10.7.1-digiumphones
10.8.0
10.8.0-digiumphones
10.8.0-digiumphones-rc1
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10.9.0
10.9.0-digiumphones
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11.0.0
11.0.0-beta1
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11.0.1
11.0.2
11.1.0
11.1.0-rc1
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11.1.1
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11.10.0
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11.11.0
11.11.0-rc1
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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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11.15.1
11.16.0
11.16.0-rc1
11.17.0
11.17.0-rc1
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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
11.20.0-rc2
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
11.3.0-rc1
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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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12.0.0
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12.3.0
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12.5.1
12.6.0
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12.7.0
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12.7.1
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12.8.0
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12.8.1
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13.0.0
13.0.0-beta1
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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
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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
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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
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certified/1.8.11-cert4
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certified/1.8.11-cert5-rc1
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certified/1.8.11-cert6
certified/1.8.11-cert7
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certified/1.8.15-cert1
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certified/1.8.15-cert4
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certified/1.8.15-cert6
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certified/1.8.28-cert1
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certified/1.8.28-cert2
certified/1.8.28-cert3
certified/1.8.28-cert4
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certified/1.8.6-cert1
certified/11.2-cert1
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certified/11.2-cert2
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certified/11.6-cert1
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certified/11.6-cert10
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certified/13.1-cert1
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certified/13.1-cert3
${ noResults }
3432 Commits (831af9fbc7ff3e22cc4a67f254ec5b44d6239b8a)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
69f19a5225 |
udptl: Ensure fec[] in udptl_build_packet() is initialized.
Scan results indicated that this array could be used uninitialized. At a quick look, it looks correct. In any case, initializing it is a Good Thing (tm). ........ Merged revisions 359457 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359458 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359459 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
9b31bd3cd8 |
Fix deadlock potential with some ast_indicate/ast_indicate_data calls.
Calling ast_indicate()/ast_indicate_data() with the channel lock held can result in a deadlock with a local channel because of how local channels need to avoid deadlock. ........ Merged revisions 359451 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359453 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359455 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
40289b63db |
Fix incorrect jitter buffer overflow due to missed resynchronizations
When a change in time occurs, such that the timestamps associated with frames being placed into an adaptive jitter buffer (implemented in jitterbuf.c) are significantly different then the previously inserted frames, the jitter buffer checks to see if it needs to be resynched to the new time frame. If three consecutive packets break the threshold, the jitter buffer resynchs itself to the new timestamps. This currently only occurs when history is calculated, and hence only on JB_TYPE_VOICE frames. JB_TYPE_CONTROL frames, on the other hand, are never passed to the history calculations. Because of this, if the jump in time is greater then the maximum allowed length of the jitter buffer, the JB_TYPE_CONTROL frames are dropped and no resynchronization occurs. Alterntively, if the overfill logic is not triggered, the JB_TYPE_CONTROL frame will be placed into the buffer, but with a time reference that is not applicable. Subsequent JB_TYPE_VOICE frames will quickly trigger the overflow logic until reads from the jitter buffer reach the errant JB_TYPE_CONTROL frame. This patch allows JB_TYPE_CONTROL frames to resynch the jitter buffer. As JB_TYPE_CONTROL frames are unlikely to occur in multiples, it perform the resynchronization on any JB_TYPE_CONTROL frame that breaks the resynch threshold. Note that this only impacts chan_iax2, as other consumers of the adaptive jitter buffer use the abstract jitter buffer API, which does not use JB_TYPE_CONTROL frames. Review: https://reviewboard.asterisk.org/r/1814/ (closes issue ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt Jordan Patches: jitterbuffer-2012-2-26.diff uploaded by Kris Shaw (license 5722) ........ Merged revisions 359356 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359358 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359359 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
2019a7e6b9 |
Fix Dial m and r options and forked calls generating warnings for voice frames.
When connected line support was added, the wait_for_answer() variable single changed its meaning slightly. Unfortunately, the places where single was used did not necessarily get updated to reflect that change. Also audio/video frames were sent to all forked calls when the endpoints were never made compatible. * Don't pass audio/video media frames when the channels have not been made compatible. * Added handling of AST_CONTROL_SRCCHANGE to app_dial.c. * Fixed app_dial.c passing on AST_CONTROL_HOLD because that frame can also pass a requested MOH class. (closes issue ASTERISK-16901) Reported by: Chris Gentle (closes issue ASTERISK-17541) Reported by: clint Review: https://reviewboard.asterisk.org/r/1805/ ........ Merged revisions 359344 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359355 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359357 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
00b270833f |
Fix bogus reads/writes of console log levels in asterisk.c
This patch updates the NUMLOGLEVELS define in logger.h to 32, to match the fact that logger.c implements 32 log levels (because of the custom log level stuff). asterisk.c uses this define to size an array of levels per remote console. This array is modified in ast_console_toggle_loglevel(), which is called by the "logger set level" CLI command. While the documentation for the CLI command doesn't make it terribly obvious, you can use this CLI command to toggle a custom log level on a remote console, as well. However, doing so led to an invalid array index in asterisk.c. This array is read from any time a log message is written to a console. So, all custom log level messages resulted in a bogus read if a remote console was connected. ........ Merged revisions 359259 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359260 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359261 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
6ac425df31 |
Fix inaccurate sizeof() in sched.c.
This code just needed sizeof(int), not sizeof(int *). ........ Merged revisions 359157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359162 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359166 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
9410f85699 |
Fix incorrect sizeof() usage in features.c.
This didn't actually result in a bug anywhere, luckily. The only place where the result of these memcpys was used is in app_dial, and the only field that it read out of ast_call_feature was the first one, which is an int, so these memcpys always copied just enough to avoid a problem. ........ Merged revisions 359069 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359072 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359075 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
1b3cbdacd7 |
Fix incorrect sizeof() on a pointer in MD5Final().
........ Merged revisions 359059 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359060 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359061 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
6ec5c103d6 |
Don't use a buffer after it goes out of scope.
's' is set to 'workspace'. Make sure 'workspace' doesn't go out of scope while the reference to it via 's' is still used. ........ Merged revisions 359056 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359057 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359058 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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cb94c35a85 |
Fix setting CDR variables in the hangup extension
A previous CDR fix for setting CDR variables during a bridge via custom dialplan features broke setting CDR variables in the hangup extension. This patch fixes the issue. Review: https://reviewboard.asterisk.org/r/1794/ ........ Merged revisions 358978 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358989 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358993 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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699d2bd705 |
Make hints for invalid SIP devices return Unavail, not idle
This patch drastically simplifies the device state aggegation code. The old method was not only overly complex, but also made it impossible to return AST_DEVICE_INVALID from the aggregation code. The unit test update is as a result of fixing that bug. The SIP change stems from a bug introduced by removing a DNS lookup for hostname-based SIP channels. (closes issue ASTERISK-16702) Review: https://reviewboard.asterisk.org/r/1808/ ........ Merged revisions 358943 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358944 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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786f5898d1 |
Finalize ast_channel opaquification
Review: https://reviewboard.asterisk.org/r/1786/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358907 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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73ec67e008 |
Fix crash caused by opaquification change -r356042.
The set_format() function was more subtle in how it modified the struct ast_channel readtrans/writetrans values. * Fixed ast_activate_generator() conversion correctly. (closes issue ASTERISK-19434) Reported by: Birger Harzenetter Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358861 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c7315c4283 |
Use struct copy instead of memcpy().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358858 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f5fda0eb74 |
Transition app_page to using app_confbridge internally for the conference bridge portion of paging. This also adds a new 'announcement' option to ConfBridge user profiles.
Review: https://reviewboard.asterisk.org/r/1754/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358730 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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11bbc097b7 |
Eliminate double close of file descriptor in manager.c
The process_output function in manager.c attempted to call fclose and close immediately afterwards. Since fclose implies close, this resulted in a potential double free on file descriptors. This patch changes that behavior and also adds error checking to fclose and close depending on which was deemed necessary. Also error messages. Thanks to Rosen Iliev for pointing out the location of the problem. (closes issue ASTERISK-18453) Reported By: Jaco Kroon Review: https://reviewboard.asterisk.org/r/1793/ ........ Merged revisions 358214 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358215 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358216 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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8d1bde49a9 |
Fix case-sensitivity for device-specific event subscriptions and CCSS
This change fixes case-sensitivity for device-specific subscriptions such that the technology identifier is case-insensitive while the remainder of the device string is still case-sensitive. This should also preserve the original case of the device string as passed in to the event system. CCSS is the only feature affected as it is the only consumer of device-specific event subscriptions. The second part of this patch addresses similar case-sensitivity issues within CCSS itself that prevented it from functioning correctly after the fix to the events system. This adds a unit test to verify that the event system works as expected. (closes issue ASTERISK-19422) Review: https://reviewboard.asterisk.org/r/1780/ ........ Merged revisions 357940 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357941 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357942 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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9926662aba |
Remove ISDN hold restriction for non-bridged calls.
The check if an ISDN call is bridged before it could be placed on hold is not necessary and is overly restrictive. The check was originally done to prevent problems with call transfers in case a user tried to transfer a call connected to an application to another call connected to an application. The ISDN transfer code has not required this restriction for quite some time because ECT could transfer any two active calls to each other. * Remove ISDN hold restriction for calls connected to applications. * Made ast_waitfordigit_full() ignore AST_CONTROL_HOLD and AST_CONTROL_UNHOLD instead of generating a warning message. (closes issue ASTERISK-19388) Reported by: Birger Harzenetter Tested by: rmudgett ........ Merged revisions 357894 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357895 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357896 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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fc558d28f2 |
Fix race condition that can cause important control frames (such as a hangup) to be missed.
This takes two actions. 1. Move the reading of the alertpipe in __ast_read() to immediately before the removal of frames from the readq. This means we won't do something silly like read from the alertpipe, then ignore the fact that there's a frame to get from the readq since channel's fdno is the AST_TIMING_FD. 2. When ast_settimeout() sets the rate to 0 and the timingfunc to NULL, if the channel's fdno is the AST_TIMING_FD, then set the fdno to -1. This is because if the rate is 0 and the timingfunc is NULL, it means that the channel's timing fd is being invalidated, so any pending reads should not occur. This may actually solve more issues than the referenced one below, but it's not known at this time for sure. (closes issue ASTERISK-19223) reported by Frank-Michael Wittig Review: https://reviewboard.asterisk.org/r/1779 ........ Merged revisions 357761 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357762 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357775 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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0e5c761c28 |
Opaquify ast_channel typedefs, fd arrays, and softhangup flag
Review: https://reviewboard.asterisk.org/r/1784/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e291318df2 |
Prevent outbound SIP NOTIFY packets from displaying a port of 0
In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which changed the behavior of ast_find_ourip such that port number was wiped out. This caused the port in internip (which is used for Contact and Call-ID on NOTIFYs) to be 0. This change causes ast_find_ourip to be port-preserving again. (closes issue ASTERISK-19430) ........ Merged revisions 357665 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357667 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357673 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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41f5a1ab35 |
Update stringfield documentation for removed second va_list in favor of va_copy.
In r320946, the second va_list that was passed to ast_string_field_build_va and friends, was removed. This patch updates the documentation to reflect that. ........ Merged revisions 357620 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357621 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a9d607a357 |
Opaquify ast_channel structs and lists
Review: https://reviewboard.asterisk.org/r/1773/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e063fa6b3f |
Fix REF_DEBUG compile errors.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357404 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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85ea4277f1 |
Convert struct ast_tcptls_session_instance to finally use the ao2 object lock.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357317 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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2e834f7d36 |
Astobj2 locking enhancement.
Add the ability to specify what kind of locking an ao2 object has when it is allocated. The locking could be one of: MUTEX, RWLOCK, or none. New API: ao2_t_alloc_options() ao2_alloc_options() ao2_t_container_alloc_options() ao2_container_alloc_options() ao2_rdlock() ao2_wrlock() ao2_tryrdlock() ao2_trywrlock() The OBJ_NOLOCK and AO2_ITERATOR_DONTLOCK flags have a slight meaning change. They no longer mean that the object is protected by an external mechanism. They mean the lock associated with the object has already been manually obtained by one of the ao2_lock calls. This change is necessary for RWLOCK support since they are not reentrant. Also an operation on an ao2 container may require promoting a read lock to a write lock by releasing the already held read lock to re-acquire as a write lock. Replaced API calls: ao2_t_link_nolock() ao2_link_nolock() ao2_t_unlink_nolock() ao2_unlink_nolock() with the respective ao2_t_link_flags() ao2_link_flags() ao2_t_unlink_flags() ao2_unlink_flags() API calls to be more flexible and to allow an anticipated enhancement to control linking duplicate objects into a container. The changes to format.c and format_cap.c are taking advantange of the new ao2 locking options to simplify the use of the format capabilities containers. Review: https://reviewboard.asterisk.org/r/1554/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357272 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5b821af99a |
Trailing whitespace cleanup.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357178 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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50c8557f03 |
Add ability to clone ao2 containers.
Occasionally there is a need to put all objects in one container also into another container. Some reasons you might need to do this: 1) You need to reconfigure a container. You would do this by creating a new container with the new configuration and ao2_container_dup the old container into it. Then replace the old container with the new. Then destroy the old container. 2) You need the contents of a container to remain stable while operating on all of the objects. You would do this by creating a cloned container of the original with ao2_container_clone. The cloned container is a snapshot of the objects at the time of the cloning. When done, just destroy the cloned container. Review: https://reviewboard.asterisk.org/r/1746/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357145 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ae07610d73 |
Fix ast_channel allocation init setting priority to -1 instead of 1.
* Fix opaquification conversion error.
(closes issue ASTERISK-19424)
Reported by: Jeremy Pepper
Patches:
asterisk-19424-initialize_priority_regression.diff (license #5026) patch uploaded by Michael L. Young
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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15 years ago |
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890717f305 |
Fix callerid of Originated calls.
Thanks to Matt Riddell for tracking this down. (closes issue ASTERISK-19385) Reported by: ornix ........ Merged revisions 357093 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357095 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357096 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c20cfcdcf0 |
Address comments from Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357014 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1fac2fba4b |
Deprecated macro usage for connected line, redirecting, and CCSS
This commit adds GoSub alternatives to connected line, redirecting, and CCSS macro hooks so that macro can finally be deprecated. This also adds deprecation warnings for those features when used and in documentation. Review: https://reviewboard.asterisk.org/r/1760/ (closes issue SWP-4256) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357013 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3cf09f40f7 |
Convert netsock.h over to use ast_sockaddrs rather than sockaddr_in and update
chan_iax2 to pass in the correct types. chan_iax2 is the only consumer for the various ast_netsock_* functions in trunk at this point, so this feels like a safe change to make. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357005 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a9f4d13b02 |
Copy CDR variables when set during a bridge
This patch makes sure amaflags, accountcode, and userfield get copied to the bridge CDR when set during a bridge (like via a custom feature). (closes issue ASTERISK-16990) Review: https://reviewboard.asterisk.org/r/1721/ ........ Merged revisions 356963 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356964 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356965 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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6214285950 |
Make ast_netsock_set_qos() delegate to ast_set_qos().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356916 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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51c24c88a1 |
Prefer ast_set_qos() over ast_netsock_set_qos()
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356882 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a2286c0889 |
Remove trailing whitespace
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356881 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e43d123f11 |
astobj2.h documentation updates.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356734 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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670797e5da |
Allow SRTP policies to be reloaded
Currently, when using res_srtp, once the SRTP policy has been added to the current session the policy is locked into place. Any attempt to replace an existing policy, which would be needed if the remote endpoint negotiated a new cryptographic key, is instead rejected in res_srtp. This happens in particular in transfer scenarios, where the endpoint that Asterisk is communicating with changes but uses the same RTP session. This patch modifies res_srtp to allow remote and local policies to be reloaded in the underlying SRTP library. From the perspective of users of the SRTP API, the only change is that the adding of remote and local policies are now added in a single method call, whereas they previously were added separately. This was changed to account for the differences in handling remote and local policies in libsrtp. Review: https://reviewboard.asterisk.org/r/1741/ (closes issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283) (with some small modifications for this check-in) ........ Merged revisions 356604 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356605 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356606 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ebaf59a656 |
Opaquification for ast_format structs in struct ast_channel
Review: https://reviewboard.asterisk.org/r/1770/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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235f88d122 |
Fix blind transfer parking issues if the dialed extension is not recognized as a parking extension.
Custom parking extensions may not be coded such that the first and only extension priority is the Park application. These custom parking extensions will not be recognized as parking extensions. When a call is blind transferred to an extension that is not recognized as a parking extension, the normal blind transfer code causes the transferred channel to start executing dialplan. Calls that get parked in this manner do not know the original channel name that parked the call so the original parker could never be called back if the parked call is not retrieved before the timeout time. The parking space is also announced to the call being parked as a side effect of not knowing the original parking channel. * Fix handling of BLINDTRANSFER channel variable for call parking. * Fixed SIP blind transfer using the wrong dialplan context variable to check for the parking extension. (closes issue ASTERISK-19322) Reported by: aragon Tested by: rmudgett, jparker Review: https://reviewboard.asterisk.org/r/1730/ JIRA AST-766 ........ Merged revisions 356521 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356522 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356523 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5b0f29d710 |
Revert some apparently accidental spacing changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356366 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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0cc38858dd |
Track module use count for res_calendar
If the res_calendar module was followed immediately by one of the calendar tech modules and "core stop gracefully" was run, Asterisk would crash. This patch adds use count tracking for res_calendar so that it is unloaded after the tech modules when shutting down gracefully. It is now not possible to unload all the of the calendar modules via "module unload res_calednar.so", but it is still possible to unload them all via "module unload -h res_calendar.so". Review: https://reviewboard.asterisk.org/r/1752/ ........ Merged revisions 356291 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356297 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356314 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3a9ac7c10c |
Rename ast_channel_emulate_dtmf_digit* funcs
The accessors names for the "emulate_dtmf_digit" field on the ast_channel are misleading. Change them to ast_channel_dtmf_digit_to_emulate*. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356183 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c25a442dfb |
Fix some opaquification-related compiler warnings
(closes issue ASTERISK-19419) PseudoReview - seanbright on IRC git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356152 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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4585ec1bbf |
Add missing newline to ccss state change notification
Move along, nothing to see here... ........ Merged revisions 356074 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356075 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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57f42bd74f |
ast_channel opaquification of pointers and integral types
Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a93fbe2ad5 |
Non-verbose output should always go to the remote console, regardless of the previous level.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355749 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b80fcd77e5 |
Revert a change to audio_audiohook_write_list that had no affect.
When I made this change initially, I was under the false impression that the audiohooks structure remained on the channel after all of the hooks had been detached. This is not the case, ast ast_read takes care of removing the audiohooks structure if the lists are empty. ........ Merged revisions 355622 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355623 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355624 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a78b0af5ea |
Re-commit the verbose branch.
This change permits each verbose destination (consoles, logger) to have its own concept of what the verbosity level is. The big feature here is that the logger will now be able to capture a particular verbosity level without condemning each console to need to suffer that level of verbosity. Additionally, a stray 'core set verbose' will no longer change what will go to the log. Review: https://reviewboard.asterisk.org/r/1599/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355413 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |