mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
13991 Commits (74577dd0911a4aa1ccdce06fd9faa1c64ce2de41)
Author | SHA1 | Message | Date |
---|---|---|---|
|
753f7086cd |
Merge changes from team/russell/iax2_frame_queue
This patch is an optimization for chan_iax2. This module is now heavily multi-threaded. However, there is still a good number of globally shared resources that prevent things from happen asynchronously. One of those things was the global IAX frame queue. This queue was used to hold frames that have been deferred for transmitting by another thread, and frames that may need to get retransmitted. I changed the frame queue to be per-call, since almost all of the frame queue handling only cares about frames specific to a call number. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89887 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
63bca744a2 |
Merged revisions 89844 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89844 | russell | 2007-11-27 17:21:13 -0600 (Tue, 27 Nov 2007) | 3 lines Instead of depending on the return value of ast_true(), explicitly set the eventwhencalled variable to 1. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89847 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
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 |
|
ba7f5fec38 |
Merged revisions 89837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89837 | mmichelson | 2007-11-27 17:10:05 -0600 (Tue, 27 Nov 2007) | 12 lines Two changes with regards to the 'eventwhencalled' option of queues.conf 1) Due to some signed vs. unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes' did exactly the same thing. Thus the sign change of the ast_true call. 2) The vars2manager function overwrote a \n for every channel variable it parsed, resulting in bizarre output for the channel variables. This patch remedies this. (related to issue #11385, however I'm not sure if this will actually be enough to close it) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89838 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
d15dfabe93 |
Bring in a small change from team/russell/chan_refcount
This replaces tab completion code with the use of a public function that does the same thing git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89835 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
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 |
|
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 |
|
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 |
|
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 |
|
4bf40290ae |
More "moremanager" changes - doxygen docs and changing manager version (finally)
before making more dramatic changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89770 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
28257e4005 |
More additions from the "moremanager" branch, this time for IAX2.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89769 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
989b25562f |
Blocked revisions 89727 via svnmerge
........ r89727 | mmichelson | 2007-11-27 14:22:59 -0600 (Tue, 27 Nov 2007) | 6 lines Changing some calls from free() to ast_free() since they were allocated with ast_calloc(). (closes issue #11390, reported and patched by Laureano) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89733 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
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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062327c960 |
remove a duplicate manager event
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89710 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d2b29df4f0 |
Manager events from the "moremanager" branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89706 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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09e1c572d8 |
Starting to merge changes from the "moremanager" branch. Documentation will
follow. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89702 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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35ecd08b46 |
Add an S_COR macro, which is similar to the existing S_OR macro,
except with an additional boolean arg. A hack such as: foo ? S_OR(bar, "baz") : "baz" becomes: S_COR(foo, bar, "baz") git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89683 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ab92fc1f35 |
made AEL 8-bit transparent; mainly the lexer was tossing chars with the hi-order bit set. Not nice. Also, allow @ in extension names, and a backslash, also.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89682 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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df1689e927 |
Merged revisions 89634 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89634 | russell | 2007-11-27 10:12:33 -0600 (Tue, 27 Nov 2007) | 3 lines Add a note to the sample voicemail config noting that when using IMAP storage, only the first format specified will be attached to the message. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89635 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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fd06a58800 |
Merged revisions 89631 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89631 | tilghman | 2007-11-27 09:38:03 -0600 (Tue, 27 Nov 2007) | 3 lines Default result of STAT should be "0" not "". Reported via the -users mailing list, fixed by me. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89632 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b1c0c67e76 |
Merged revisions 89624 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89624 | oej | 2007-11-27 08:34:19 +0100 (Tis, 27 Nov 2007) | 6 lines Clarify limitonpeers=yes (closes issue #11304) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89625 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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180c3fb60e |
Change all instances of "CALLERID(number)" to "CALLERID(num)" for
consistency's sake (closes issue #11381, reported and patched by jon) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89621 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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5f3a28e588 |
Merged revisions 89618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89618 | mmichelson | 2007-11-26 17:10:49 -0600 (Mon, 26 Nov 2007) | 7 lines After issuing a "say load new", if a caller hangs up during the middle of playback of a number, app_playback will continue to try to play the remaining files. With this change, no more files will be played back upon hangup. (closes issue #11345, reported and patched by IgorG) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89619 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0fe71dbb1a |
Blocked revisions 89616 via svnmerge
........ r89616 | mmichelson | 2007-11-26 17:02:30 -0600 (Mon, 26 Nov 2007) | 5 lines After issuing a "say load new" tons of warning messages are printed out to the CLI every time do_say in app_playback is called. Removing these warnings ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89617 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9bd421dd10 |
Update the configure script check for libpri to check for the newest function
that was just added. Cresl1n, please keep this in mind when making these changes to libpri or libss7. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89615 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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11df6a9119 |
Rename "limitonpeer" to "counteronpeer" since the call-limit is deprecated.
Both still works in this version. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89613 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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5070d10864 |
Formatting, doxygenification
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89611 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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96ad455115 |
Formatting changes, cleaning up some code
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89609 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d4863bb0f0 |
Start using Doxygen groupings to group variables and defines.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89607 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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0619fb1248 |
Perform some module use counting audits. This is now done outside the scope of the application/dialplan function so they do not need to worry about it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89602 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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23eecf8c61 |
Merged revisions 89599 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89599 | file | 2007-11-26 14:02:56 -0400 (Mon, 26 Nov 2007) | 6 lines Add module counting removal for error conditions. (closes issue #11333) Reported by: Laureano Patches: res_features_v2.c.patch uploaded by Laureano (license 265) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89600 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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4c992507f4 |
closes issue #11341; made changes to make utils again right with the MTX_PROFILE world.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89595 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9910b7d9a7 |
Merged revisions 89592 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89592 | file | 2007-11-26 13:36:45 -0400 (Mon, 26 Nov 2007) | 6 lines Use ast_free to free memory, or else we shall implode if MALLOC_DEBUG is enabled. (closes issue #11347) Reported by: ys Patches: pbx.pbx_config.c.diff uploaded by ys (license 281) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89593 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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b1a8de2d40 |
make sure we check to see if the configure script has been executed on a new checkout or after a distclean
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89590 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9905034266 |
Merged revisions 89587 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89587 | file | 2007-11-26 13:20:58 -0400 (Mon, 26 Nov 2007) | 6 lines Close the audio file before sending it to the post processing application. (closes issue #11357) Reported by: reformed Patches: mixmonitor.patch uploaded by reformed (license 330) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89589 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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0b120dac62 |
Merged revisions 89580 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89580 | mmichelson | 2007-11-26 09:48:06 -0600 (Mon, 26 Nov 2007) | 6 lines Revert vmu->email back to an empty string if it was empty when imap_store_file was called. This prevents sending a duplicate e-mail. (closes issue #11204, reported by spditner, patched by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89581 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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71c602a2d1 |
Instead of printing out one codec in sip show channels print out all of the native ones (this is for video).
(closes issue #11366) Reported by: ovi git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89573 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |