mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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10.10.0
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10.12.2-digiumphones
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10.5.1-digiumphones
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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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11.1.0
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11.1.1
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11.10.0
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11.11.0
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11.14.0
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11.15.0
11.15.0-rc1
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11.16.0
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11.2.0
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11.2.1
11.2.2
11.20.0-rc1
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11.21.0
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11.21.1
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11.3.0
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12.0.0
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13.0.0
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certified/1.8.11-cert1
certified/1.8.11-cert10
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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/1.8.11-cert9-rc1
certified/1.8.15-cert1
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certified/1.8.28-cert1
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certified/1.8.6-cert1
certified/11.2-cert1
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certified/11.6-cert1
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certified/11.6-cert10
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certified/11.6-cert6
certified/11.6-cert7
certified/11.6-cert8
certified/11.6-cert9
certified/13.1-cert1
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certified/13.1-cert1-rc2
certified/13.1-cert1-rc3
certified/13.1-cert2
certified/13.1-cert3
${ noResults }
1229 Commits (f82c42a22e6a0883c8b51b269d3be9c50e4cd6be)
Author | SHA1 | Message | Date |
---|---|---|---|
|
ce2f670228 |
Change cdr_manager to use a "CDR" level, rather than the (overcrowded) "call" level.
(Closes issue #11015) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91173 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
ec846d6cf2 |
Merged revisions 91074 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91074 | russell | 2007-12-04 18:48:47 -0600 (Tue, 04 Dec 2007) | 4 lines When DEBUG_THREADS is enabled, we only have the details about who is holding a lock that we are waiting on for a mutex, not rwlocks. This should fix the problem where people have reported "core show locks" crashing sometimes. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91077 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
f52c068605 |
Merged revisions 90967 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90967 | russell | 2007-12-04 13:57:39 -0600 (Tue, 04 Dec 2007) | 7 lines Make some changes to some additions I made recently for doing channel autoservice when looking up extensions. This code was added to handle the case where a dialplan switch was in use that could block for a long time. However, the way that I added it, it did this for all extension lookups. However, lookups in the in-memory tree of extensions should _not_ take long enough to matter. So, move the autoservice stuff to be only around executing a switch. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91011 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
f15be28fb0 |
Add support for monitoring MWI on FXO lines.
This introduces two new options for zapata.conf: mwimonitor and mwimonitornotify. The mwimonitor option enables MWI monitoring. When the MWI state on a line changes, then the script specified by mwimonitornotify will be executed for custom handling of the state change, similar to the externnotify option of voicemail.conf. Also, when the MWI state on an FXO line changes, an internal Asterisk event is generated to indicate the new state of the associated mailbox. That may, any module that cares about MWI information will get notified and can handle it just as if app_voicemail had sent this notification. (BE-253, original patch from markster, with some minor modifications by me to add comments, documentation, and internal event support) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90949 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
cf6e8ae77e |
Fix build, that some people aren't seeing for some reason.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90927 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
0ef0259302 |
Merged revisions 90876 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11454) ........ r90876 | qwell | 2007-12-04 11:28:08 -0600 (Tue, 04 Dec 2007) | 4 lines If we fail to create a channel after allocating a timing fd, we need to make sure to close it. Issue 11454, patch by eliel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90879 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
adf3b12e55 |
Fix a silly little typo :)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90878 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
c52d8a1cd5 |
Merged revisions 90735 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines A big one... This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop. This is accomplished by creating a datastore on the calling channel which has a linked list of all devices dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore is detached from the channel and destroyed. This change also introduces some side effects to the code which I shall enumerate here: 1. Datastore inheritance has been backported from trunk into 1.4 2. A large chunk of code has been removed from app_dial. This chunk is the section of code which handles the call forward case after the channel has been requested but before it has been called. This was removed because call-forwarding still works fine without it, it makes the code less error-prone should it need changing, and it made this set of changes much less painful to just have the forwarding handled in one place in each module. 3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore which is attached to the channel may be created and attached in either app_dial or app_queue, so they need a common place to find the datastore info. This approach was taken in case similar datastores are needed in the future, there will be a common place to add them. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
25cbb792b9 |
(closes issue #11422)
Reported by: eliel Patches: core.show.hint.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90853 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
72cafad6a2 |
A small fix from snuffy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90746 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
0147028b51 |
Avoid an additional function call. Reported by: eliel Patch by: eliel (Closes issue #11438)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90591 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
985c9f5cfe |
Merged revisions 90588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90588 | file | 2007-12-03 16:05:42 -0400 (Mon, 03 Dec 2007) | 2 lines Do not create a smoother for G723.1 frames, they need to be left alone to their native 20/24 byte size. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90589 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
46d2c050c5 |
Merged revisions 90548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90548 | file | 2007-12-03 14:40:56 -0400 (Mon, 03 Dec 2007) | 2 lines Preserve the indication currently playing on a channel when a masquerade operation happens. (issue #BE-88) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90550 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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87492f6dd6 |
Merged revisions 90432 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90432 | tilghman | 2007-12-02 03:34:23 -0600 (Sun, 02 Dec 2007) | 7 lines Clarify the return value on autoservice. Specifically, if you started autoservice and autoservice was already on, it would erroneously return an error. Reported by: adiemus Patch by: dimas (Closes issue #11433) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90433 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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6b08c442c7 |
Adding support for the "automixmonitor" dial and queue options.
This works in much the same way as the automonitor, except that instead of using the monitor app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor. This patch also introduces some new API calls to the audiohooks code for searching for an audiohook by type and for searching for a running audiohook by type. Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to be committed. (closes issue #10185, reported and patched by xmarksthespot) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90388 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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fac7480820 |
Merged revisions 90348 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90348 | russell | 2007-11-30 13:26:04 -0600 (Fri, 30 Nov 2007) | 8 lines Change the behavior of ao2_link(). Previously, in inherited a reference. Now, it automatically increases the reference count to reflect the reference that is now held by the container. This was done to be more consistent with ao2_unlink(), which automatically releases the reference held by the container. It also makes it so it is no longer possible for a pointer to be invalid after ao2_link() returns. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90351 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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afac5cce50 |
Merged revisions 90155 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90155 | tilghman | 2007-11-29 11:29:59 -0600 (Thu, 29 Nov 2007) | 5 lines Use of "private" as a field name in a header file messes with C++ projects Reported by: chewbacca Patch by: casper (Closes issue #11401) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90158 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c13ed407db |
restore behavior from the 1.4 branch... manager users created via users.conf should default to *all* permissions, not none
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90149 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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53a5f22849 |
Merged revisions 90145 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90145 | russell | 2007-11-28 18:20:34 -0600 (Wed, 28 Nov 2007) | 5 lines This set of changes is to make some callerID handling thread-safe. The ast_set_callerid() function needed to lock the channel. Also, the handlers for the CALLERID() dialplan function needed to lock the channel when reading or writing callerid values directly on the channel structure. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90146 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8a3e984b0c |
Merged revisions 90142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90142 | russell | 2007-11-28 18:06:08 -0600 (Wed, 28 Nov 2007) | 4 lines Merge a change from team/russell/chan_refcount ... This makes ast_stopstream() thread-safe. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90143 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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57c2bcca86 |
Merged revisions 90098 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90098 | kpfleming | 2007-11-28 16:30:46 -0600 (Wed, 28 Nov 2007) | 2 lines it is impossible to set permissions for manager accounts created by users.conf (reported internally, patched by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90100 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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cbe59ac93e |
file says... compile before you commit!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90099 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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e3c122863d |
Removing a pointless check of option_debug
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90061 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0b3ab17bfa |
Merged revisions 90059 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90059 | mmichelson | 2007-11-28 16:08:50 -0600 (Wed, 28 Nov 2007) | 13 lines Removing some seemingly pointless code. This sets a channel variable for every priority executed in the dialplan if you have debug set to anything non-zero. This seems pointless due to the fact that these channel variables are not referenced anywhere else in the code and their names are esoteric enough that they would not be practical to reference in the dialplan. Plus the fact that this behavior isn't documented anywhere means that the change is not likely to cause any disruption. If anything, this may actually cause a slight performance increase if running with debug on. The motivating influence for this code change is the eventwhencalled option for queues. If set to vars, all channel variables will be output to the manager. These unnecessary channel variables make the output a lot more difficult to deal with. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90060 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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58e43103d8 |
Made expr2 parser 8-bit transparent
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90039 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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aa0f63d65f |
Remove "old"-style CLI handler, since nothing uses it anymore.
Closes issue #11403, patch by eliel. This also completes the janitor project. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90038 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a6ddf3c7b0 |
Hide CLI commands starting with _ from tab completion as was done previously.
(closes issue #11395) Reported by: eliel Patches: cli.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89982 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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6aad89b97d |
Fix a few log messages.
(closes issue #11396) Reported by: IgorG Patches: spell.v1.diff uploaded by IgorG (license 20) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89981 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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880fb1ece9 |
Merged revisions 89893 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89893 | russell | 2007-11-27 18:20:13 -0600 (Tue, 27 Nov 2007) | 4 lines - update documentation for some of the goto functions to note that they handle locking the channel as needed - update ast_explicit_goto() to lock the channel as needed ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89915 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8133ad1456 |
Merged revisions 89886 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89886 | russell | 2007-11-27 17:47:28 -0600 (Tue, 27 Nov 2007) | 2 lines Don't do frame processing if ast_read() returned NULL. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89888 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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79bc5ede5f |
Merged revisions 89839 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89839 | russell | 2007-11-27 17:16:00 -0600 (Tue, 27 Nov 2007) | 2 lines Don't start/stop autoservice in pbx_extension_helper() unless a channel exists ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89840 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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1f792f1745 |
closes issue #11294; missed the conditional unlock of the contexts when the hash table is used instead; also, used the ast_free_ptr as advised.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89792 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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21a22fa584 |
Merged revisions 89790 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89790 | russell | 2007-11-27 15:45:51 -0600 (Tue, 27 Nov 2007) | 41 lines Merge changes from team/russell/autoservice_1.4 This set of changes fixes an issue that was reported to me on IRC yesterday. The user, d1mas, was using chan_zap for incoming calls and was having DTMF recognition issues in some situations. Specifically, he noticed that the problem occurred when using DISA or WaitExten. He also noticed that when using Read, the problem did not occur. His system also used DUNDi for dialplan lookups. So, he theorized that if the DUNDi lookups blocked for some period of time, that audio from the zap channel could get lost. If the audio got lost, then it wouldn't be run through the DTMF detector, and digits could get lost. He was correct, and the following set of changes fixes the problem. However, the changes go a little bit further than what was necessary to fix this exact problem. 1) I updated pbx_extension_helper() to autoservice the associated channel to handle cases where extension lookups may take a long time. This would normally be a dialplan switch that does some lookup over the network, such as the DUNDi or IAX2 switches. This ensures that even while a DUNDi lookup is blocking, the channel will be continuously serviced. 2) I made a change to the autoservice code. This is actually something that has bothered me for a long time. When a channel is in autoservice, _all_ frames get thrown away. However, some frames really shouldn't be thrown away. The most notable examples are signalling (CONTROL) frames, and DTMF. So, this patch queues up important frames while a channel is in autoservice. When autoservice is stopped on the channel, the queued up frames get stuck back on the channel so that they can get processed instead of thrown away. 3) I made another change to the autoservice code to handle the case where autoservice is started on channels recursively. Previously, you could call ast_autoservice_start() multiple times on a channel, and it would stop the first time ast_autoservice_stop() gets called. Now, it will ensure that autoservice doesn't actually stop until the final call to ast_autoservice_stop(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89791 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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5ac0f923b3 |
A few more "moremanager" fixes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89772 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a30972ee65 |
More "moremanager" fixes. Manager commands to check module status.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89771 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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5e6efa075a |
Merged revisions 89709 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89709 | kpfleming | 2007-11-27 14:16:56 -0600 (Tue, 27 Nov 2007) | 2 lines on second thought... revert all the other changes i've made in app options parsing leaving only one: if an empty argument is supplied for an option, set that argument pointer to point to an empty string rather than NULL, so that the application can do normal checks on it without worrying about it being NULL ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8ecda22e06 |
Merged revisions 89701 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89701 | kpfleming | 2007-11-27 13:36:55 -0600 (Tue, 27 Nov 2007) | 2 lines generate a warning when an application option that requires an argument is ignored due to lack of an argument ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89704 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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df7ba90b20 |
The following patch with updates for trunk. Works much better in trunk.
Also by accident fixed a bad typo by a previous committer, which actually made video calls not work fully... Merged revisions 89630 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines If we get a codec offer using a well-known payload type, but using it for another codec that we don't know, Asterisk did not remove that codec from the list. With this patch, we remove the codec from audio and video rtp objects and deny it ever existed. Thanks to lasse for testing. (closes issue #11376) Reported by: lasse Patches: bug11376.txt uploaded by oej (license 306) Tested by: lasse ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89698 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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98fbe822f5 |
Ensure the value returned from ast_random is between 0 and RAND_MAX on 64-bit platforms.
(closes issue #11348) Reported by: sperreault git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89637 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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4d8932a6dc |
Merged revisions 89622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 line closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89623 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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89c0a0a763 |
Merged revisions 89610 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89610 | file | 2007-11-26 17:10:29 -0400 (Mon, 26 Nov 2007) | 2 lines Fix issues with async dialing with an application executing. The application has to be terminated and control returned to the thread before hanging things up. (issue #BE-252) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89612 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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130a2051fa |
- Mark "concise" as deprecated
- Restructure other changes to UPGRADE.txt and CHANGES We're still looking for scripts that replace asterisk -rx "show shannels concise" by using the manager interface, but still produces the same output. Anyone? git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89606 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b11f846e7e |
Merged revisions 89594 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89594 | russell | 2007-11-26 11:41:04 -0600 (Mon, 26 Nov 2007) | 3 lines Add channel locking to a function that needed to be doing it. This is just a little something I noticed while working on a completely unrelated issue. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89596 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d3209e2ef2 |
closes issue #11356; Many thanks to snuffy for his code review and changes to cut down duplication. I tested this against hashtest, and it passes. I reviewed the changes, and they look reasonable. I had to remove a few const decls to make things compile on my workstation,
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89591 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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721b3c8a0e |
Merged revisions 89586 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89586 | kpfleming | 2007-11-26 11:20:36 -0600 (Mon, 26 Nov 2007) | 2 lines when parsing application options that take arguments, don't indicate that the option was supplied unless a non-zero-length argument was found for it ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89588 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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2ec4b57622 |
Thanks to pnlarsson for noting the spelling error in the cli commands. Also, added some verbage about the new algorithm to CHANGES.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89583 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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72a8e08bf5 |
Revert change for 11348 until it can be looked at even more.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89582 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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867b734868 |
Merged revisions 89577 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89577 | file | 2007-11-26 11:34:38 -0400 (Mon, 26 Nov 2007) | 6 lines If channel allocation fails because the alert pipe could not be created also free the scheduler context. (closes issue #11355) Reported by: eliel Patches: main.channel.c.patch uploaded by eliel (license 64) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89578 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a1b79aa278 |
Make the behavior of using /dev/urandom for random numbers the same as random().
(closes issue #11348) Reported by: sperreault Patches: ast_random2.diff uploaded by sperreault (license 252) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89576 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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595961655a |
Try to get channel.h and channel.c aligned in regards to ast_set_callerid as well
as change name of variables to follow the rest of the naming. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89564 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |