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${ noResults }
133 Commits (573e35bba197a27e7ea15e4d5004d0898af41c38)
Author | SHA1 | Message | Date |
---|---|---|---|
|
a584b51dd9 |
Merged revisions 105570 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105570 | russell | 2008-03-03 11:16:53 -0600 (Mon, 03 Mar 2008) | 3 lines In the case of an ast_channel allocation failure, take the local_pvt out of the pvt list before destroying it. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105571 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
bc3f768547 |
Merged revisions 105568 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105568 | russell | 2008-03-03 11:05:16 -0600 (Mon, 03 Mar 2008) | 3 lines Fix a potential memory leak of the local_pvt struct when ast_channel allocation fails. Also, in passing, centralize the code necessary to destroy a local_pvt. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105569 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
b895cda7a3 |
Merged revisions 103904 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103904 | mmichelson | 2008-02-20 15:40:08 -0600 (Wed, 20 Feb 2008) | 6 lines Fix a crash if the channel becomes NULL while attempting to lock it. (closes issue #12039) Reported by: danpwi ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103908 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
85cd43c3c9 |
Merged revisions 103821 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103821 | russell | 2008-02-19 14:02:49 -0600 (Tue, 19 Feb 2008) | 8 lines Account for the fact that the "other" channel can disappear while the local pvt is not locked. (fixes a problem introduced in rev 100581) (closes issue #12012) Reported by: stevedavies Patch by me ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103822 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
1c74c549d7 |
Merged revisions 100581 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100581 | russell | 2008-01-28 11:15:41 -0600 (Mon, 28 Jan 2008) | 9 lines Make some deadlock related fixes. These bugs were discovered and reported internally at Digium by Steve Pitts. - Fix up chan_local to ensure that the channel lock is held before the local pvt lock. - Don't hold the channel lock when executing the timing function, as it can cause a deadlock when using chan_local. This actually changes the code back to be how it was before the change for issue #10765. But, I added some other locking that I think will prevent the problem reported there, as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100582 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
949bb30d03 |
Merged revisions 99594 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99594 | oej | 2008-01-22 18:41:57 +0100 (Tis, 22 Jan 2008) | 3 lines Add more dependencies on chan_local and add a note to the description of chan_local so that people don't disable it in menuselect just to clean up. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99596 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
c48b43e2d8 |
Merged revisions 99426 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99426 | mmichelson | 2008-01-21 17:55:26 -0600 (Mon, 21 Jan 2008) | 12 lines Fixing an issue wherein monitoring local channels was not possible. During a channel masquerade, the monitors on the two channels involved are swapped. In 99% of the cases this results in the desired effect. However, if monitoring a local channel, this caused the monitor which was on the local channel to get moved onto a channel which is immediately hung up after the masquerade has completed. By swapping the monitors prior to the masquerade, we avoid the problem by tricking the masquerade into placing the monitor back onto the channel where we want it. During the investigation of the issue, the channel's monitor was the only thing that was swapped in such a manner which did not make sense to have done. All other variable swapping made sense. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99427 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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343949b741 |
Merged revisions 98964 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98964 | mmichelson | 2008-01-16 11:20:11 -0600 (Wed, 16 Jan 2008) | 10 lines Fix a deadlock in chan_local in local_hangup. There was contention because the local_pvt was held and it was attempting to lock a channel, which is the incorrect locking order. (closes issue #11730) Reported by: UDI-Doug Patches: 11730.patch uploaded by putnopvut (license 60) Tested by: UDI-Doug ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98965 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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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 |
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7e8835e0d7 |
remove another set of redundant #include "asterisk/options.h"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0595b5e2aa |
include "logger.h" and errno.h from asterisk.h - usage shows that they
were included almost everywhere. Remove some of the instances. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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1e6489a175 |
trim more redundant headers
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89384 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d82a631f9c |
more removal of duplicate #include lines
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89349 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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fdb7f7ba3d |
Start untangling header inclusion in a way that does not affect
build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c472a68a8a |
Correctly use defined return values in (some) load_module functions.
(issue #11096) Patches: chan_agent.c.patch uploaded by eliel (license 64) chan_local.c.patch uploaded by eliel (license 64) chan_features.c.patch uploaded by eliel (license 64) chan_zap.c.patch uploaded by eliel (license 64) res_monitor.c.patch uploaded by eliel (license 64) res_realtime.c.patch uploaded by eliel (license 64) res_crypto.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87202 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ebe4050128 |
Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former didn't make much sense
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86820 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b0f3e6097e |
Convert NEW_CLI to AST_CLI.
Closes issue #11039, as suggested by seanbright. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86536 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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46b9ca721b |
Add jitterbuffer support for chan_local. To enable it, you use the 'j' option
in the Dial command. The 'j' option _must_ be used in conjunction with the 'n' option. This feature will allow you to use the existing jitterbuffer implementation to put a jitterbuffer on incoming SIP calls connecting to Asterisk applications by putting a local channel in the middle. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85097 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c7a9ec1691 |
(issue #10724)
Reported by: eliel Patches: res_features.c.patch uploaded by eliel (license 64) res_agi.c.patch uploaded by seanbright (license 71) res_musiconhold.c.patch uploaded by seanbright (license 71) pbx.c.patch uploaded by moy (license 222) logger.c.patch uploaded by moy (license 222) frame.c.patch uploaded by moy (license 222) manager.c.patch uploaded by moy (license 222) http.c.patch uploaded by moy (license 222) dnsmgr.c.patch uploaded by moy (license 222) res_realtime.c.patch uploaded by eliel (license 64) res_odbc.c.patch uploaded by seanbright (license 71) res_jabber.c.patch uploaded by eliel (license 64) chan_local.c.patch uploaded by eliel (license 64) chan_agent.c.patch uploaded by eliel (license 64) chan_alsa.c.patch uploaded by eliel (license 64) chan_features.c.patch uploaded by eliel (license 64) chan_sip.c.patch uploaded by eliel (license 64) RollUp.1.patch (includes all of the above patches) uploaded by seanbright (license 71) Convert many CLI commands to the NEW_CLI format. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82930 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ded76a8988 |
convert various places that access the channel lock directly to use the channel lock wrappers
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82728 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0e83c80480 |
Merged revisions 79902 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10485) ........ r79902 | qwell | 2007-08-17 12:44:22 -0500 (Fri, 17 Aug 2007) | 4 lines Re-add the setting of callerid name and number. Issue 10485, reported by and fix explained by paradise. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79903 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d5eda8709c |
Merged revisions 79174 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79174 | file | 2007-08-13 11:18:04 -0300 (Mon, 13 Aug 2007) | 4 lines (closes issue #10437) Reported by: haklin Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79175 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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4e0947c5f1 |
Convert code that checks the _softhangup member of ast_channel directory to use
the ast_check_hangup() funciton. This function takes scheduled hangups into account. (closes issue #10230, patch by Juggie) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77858 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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55b1ee298e |
Merge the dialplan_aesthetics branch. Most of this patch simply converts applications
using old methods of parsing arguments to using the standard macros. However, the big change is that the really old way of specifying application and arguments separated by a comma will no longer work (e.g. NoOp,foo|bar). Instead, the way that has been recommended since long before 1.0 will become the only method available (e.g. NoOp(foo,bar). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76703 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0e969271ae |
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8a7732f067 |
via 10206, I have added an option (e) to Dial to allow the h exten to get run on peer. Had to upgrade ast_flag stuff to 64 bits to do this.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75400 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a1b9cbcd31 |
Implementation of a feature that will disable "missed calls" counters on SIP phones.
If the call is answered by another phone, other phones won't display the call as "missed". You can also add an option to the dial command so that you can have a "followme" scenario and not count the calls as "missed" when you cancel the call. Thanks to Ramon and Frank for feedback on this feature. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74024 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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92de3e5939 |
Merged revisions 73319 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73319 | file | 2007-07-05 10:27:40 -0300 (Thu, 05 Jul 2007) | 10 lines Merged revisions 73318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul 2007) | 2 lines Actually check to make sure a PBX was started on one of the Local channels instead of blindly assuming it was. (issue #10112 reported by makoto) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73320 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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055d82cbce |
Add a massive set of changes for converting to use the ast_debug() macro.
(issue #9957, patches from mvanbaak, caio1982, critch, and dimas) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69327 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9d05ff8ed5 |
Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a76ff39c0e |
Merged revisions 64193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64193 | murf | 2007-05-14 07:58:42 -0600 (Mon, 14 May 2007) | 1 line As per 9570, worrisome CDR warnings have been removed, that are either not helpful, or not relevant. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64208 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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3d409eb793 |
Update the device state functionality of chan_local such that it will return
NOT_INUSE or INUSE when Local channels are in use as opposed to just UNKNOWN. It will still return INVALID if the extension doesn't exist at all. (issue #8048, patch from tim_ringenbach) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62673 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ecaf781933 |
Merged revisions 60989 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered. This also adds the mods from 1.4/r.61136; ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61152 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c6c64af72d |
Merged revisions 60847 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60847 | tilghman | 2007-04-08 21:42:48 -0500 (Sun, 08 Apr 2007) | 10 lines Merged revisions 60846 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08 Apr 2007) | 2 lines Bug 9505 - If the return value for local_queue_frame is set, then p->lock is no longer valid. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60848 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b3bc675402 |
Merged revisions 57318 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r57318 | file | 2007-03-01 17:21:44 -0500 (Thu, 01 Mar 2007) | 10 lines Merged revisions 57317 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar 2007) | 2 lines Don't even attempt to optimize things when a proxy channel is involved. It will just explode in weird and unexplaineable ways. (issue #9175 reported by clegall_proformatique) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57319 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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dcca8f345f |
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines Merge the changes from the /team/group/vldtmf_fixup branch. The main bug being addressed here is a problem introduced when two SIP channels using SIP INFO dtmf have their media directly bridged. So, when a DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk would try to emulate a digit of some length by first sending a DTMF BEGIN frame and sending a DTMF END later timed off of incoming audio. However, since there was no audio coming in, the DTMF_END was never generated. This caused DTMF based features to no longer work. To fix this, the core now knows when a channel doesn't care about DTMF BEGIN frames (such as a SIP channel sending INFO dtmf). If this is the case, then Asterisk will not emulate a digit of some length, and will instead just pass through the single DTMF END event. Channel drivers also now get passed the length of the digit to their digit_end callback. This improves SIP INFO support even further by enabling us to put the real digit duration in the INFO message instead of a hard coded 250ms. Also, for an incoming INFO message, the duration is read from the frame and passed into the core instead of just getting ignored. (issue #8597, maybe others...) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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ec8591d04c |
Constify a bunch of usage strings for CLI commands.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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f781673614 |
Merged revisions 47751 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47751 | file | 2006-11-16 13:29:12 -0500 (Thu, 16 Nov 2006) | 10 lines Merged revisions 47750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov 2006) | 2 lines Because of the way chan_local is written we should be extra careful and make sure our callback functions have a tech_pvt. (issue #8275 reported by mflorell) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47752 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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176c7ca47c |
Merged revisions 47712 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47712 | file | 2006-11-15 17:31:17 -0500 (Wed, 15 Nov 2006) | 10 lines Merged revisions 47711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov 2006) | 2 lines Make sure that the pvt structure exists before trying to do fixup on Local channels. (issue #7937 reported by mada123, fix by alamantia with mods by me) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47713 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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908f176cf3 |
A fair number of changes for the sake of bug 7506
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47290 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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7659f6d524 |
Merged revisions 47287 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47287 | file | 2006-11-07 15:14:58 -0500 (Tue, 07 Nov 2006) | 2 lines This is not the commit you are looking for... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47288 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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66d3792f11 |
Merged revisions 47284 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47284 | file | 2006-11-07 15:08:52 -0500 (Tue, 07 Nov 2006) | 2 lines Make MOH work as it did before in chan_local, without this then it can go funky when transfers and MOH are involved. (issue #7671 reported by jmls) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47285 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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10875731ec |
Merged revisions 47051 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47051 | tilghman | 2006-11-02 17:00:20 -0600 (Thu, 02 Nov 2006) | 2 lines Reverse change of "show" to "list" and make several other commands more consistent with "category verb arguments" ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47052 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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36a9656234 |
Merged revisions 43697 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43697 | file | 2006-09-26 16:19:33 -0400 (Tue, 26 Sep 2006) | 2 lines Strip options off the argument passed for devicestate in chan_local. (issue #8034 reported by pcardozo) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43698 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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6ae8741518 |
Lots more removal of deprecated things
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43452 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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fcb999c01c |
merge qwell's CLI verbification work
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43212 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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082f59ff03 |
Use flags instead of variables on the private structure for things
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42338 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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0b93665a71 |
Code cleaning/updates/potential bug fixes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42311 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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c6977b9983 |
Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41507 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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f7e7161607 |
Merge team/russell/frame_caching
There are some situations in Asterisk where ast_frame and/or iax_frame structures are rapidly allocatted and freed (at least 50 times per second for one call). This code significantly improves the performance of ast_frame_header_new(), ast_frdup(), ast_frfree(), iax_frame_new(), and iax_frame_free() by keeping a thread-local cache of these structures and using frames from the cache whenever possible instead of calling malloc/free every time. This commit also converts the ast_frame and iax_frame structures to use the linked list macros. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41278 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |