mirror of https://github.com/asterisk/asterisk
master
20
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1.4.18.1
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1.6.0
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1.8.9.0
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10.1.1
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10.10.0
10.10.0-digiumphones
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10.12.0
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10.12.1
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10.12.2-digiumphones
10.12.3
10.12.3-digiumphones
10.12.4
10.12.4-digiumphones
10.2.0
10.2.0-rc1
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10.2.1
10.3.0
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10.4.0
10.4.0-digiumphones-rc1
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10.5.0
10.5.0-digiumphones
10.5.0-digiumphones-rc1
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10.7.0-digiumphones
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10.8.0
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10.8.0-digiumphones-rc1
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10.9.0-digiumphones-rc3
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11.5.0
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11.5.1
11.6-cert11
11.6.0
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12.0.0
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13.0.1
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13.1.1
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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-cert6
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certified/1.8.28-cert1
certified/1.8.28-cert1-rc1
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certified/1.8.6-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/11.6-cert6
certified/11.6-cert7
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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 }
2393 Commits (cce4fad52281001b51f61a8c33ebe04838e8f18c)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
cce4fad522 |
Make invalid hints report Unavailable instead of Idle.
(closes issue #14413) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203702 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
59c1998d67 |
Improve T.38 negotiation by exchanging session parameters between application and channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203699 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
0264eef115 |
Merge the new Channel Event Logging (CEL) subsystem.
CEL is the new system for logging channel events. This was inspired after facing many problems trying to represent what is possible to happen to a call in Asterisk using CDR records. For more information on CEL, see the built in HTML or PDF documentation generated from the files in doc/tex/. Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard work developing this code. Also, thanks to Matt Nicholson (mnicholson) and Sean Bright (seanbright) for their assistance in the final push to get this code ready for Asterisk trunk. Review: https://reviewboard.asterisk.org/r/239/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e06c6f97c4 |
Add functions to map syslog facilities and priorities constants to strings.
Also change the default casing of the string contants to lowercase. This really just saves us from have to lowercase them later when displaying them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203605 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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2f88262abb |
Add checks in configure for non-POSIX syslog facilities.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203569 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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fe07434497 |
One more formatting nit ... use spaces for inline indentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203534 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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cd1e7b61df |
Convert spaces to tabs for indentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203525 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4535305772 |
Move syslog utility functions into a separate file so they can be re-used.
This has the pleasant side effect of cleaning up the header inclusion process in logger.c. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203508 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a6426cdf9d |
fixes a few redundant conditions
(issue #15269) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203444 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ef9b9e36d4 |
Merged revisions 203380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009) | 4 lines I didn't see that Mark already fixed the underlying issue! Yay for removing useless code. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203381 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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bd2e4dc229 |
Merged revisions 203375 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009) | 9 lines Fix a case where CDR answer time could be before the start time involving parking. (closes issue #13794) Reported by: davidw Patches: 13794.patch uploaded by murf (license 17) 13794.patch.160 uploaded by murf (license 17) Tested by: murf, dbrooks ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203376 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3814caadad |
Merged revisions 203311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203311 | twilson | 2009-06-25 15:09:15 -0500 (Thu, 25 Jun 2009) | 2 lines Don't try to free NULL ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203338 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ca3a181c33 |
Pass a logmsg to ast_log_vsyslog instead of separate arguments.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203188 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ef667f2f82 |
If we delete the info, lets also delete the lines
(closes issue #14509) Reported by: timeshell Patches: 20090504__bug14509.diff.txt uploaded by tilghman (license 14) Tested by: awk, timeshell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202753 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0bbd5c9424 |
Ignore voicemail messages that are just silence.
(closes issue #2264) Reported by: pfn Patches: silent-vm-1.6.2.txt uploaded by pfn (license 810) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202570 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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2affa3e999 |
Merged revisions 202496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009) | 4 lines Report CallerID change during a masquerade. Reported by: markster ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202497 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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87c8658912 |
attempting to load running modules
Modules placed in the priority heap for loading were not properly removed from the linked list. This resulted in some modules attempting to load twice. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202410 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b390327dd7 |
Remove unnecessary usleep() from a couple of module unload callbacks.
In passing, also tweak cdr_unregister() to hold the list lock a bit less time. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202109 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e85296e244 |
Add support for allowing an RTP engine to decide on whether it is possible for specific formats to be transcoded for an RTP instance.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201902 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ad0d1bfd9e |
Merged revisions 201828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009) | 6 lines If the "h" extension fails, give it another chance in main/pbx.c. If the "h" extension fails, give it another chance in main/pbx.c, when it returns from the bridge code. Fixes an issue where the "h" extension may occasionally not fire, when a Dial is executed from a Macro. Debugged in #asterisk with user tompaw. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201829 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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dcfe69ec64 |
fixes some memory leaks and redundant conditions
(closes issue #15269) Reported by: contactmayankjain Patches: patch.txt uploaded by contactmayankjain (license 740) memory_leak_stuff.trunk.diff uploaded by dvossel (license 671) Tested by: contactmayankjain, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201678 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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dce6a54a4a |
Trunk implementation of setting an alternate RTP source.
This contains the interface by which we can let an rtp instance know that it might start receiving audio from a new source. This is similar in nature to revision 197588 of Asterisk 1.4. Review: https://reviewboard.asterisk.org/r/276 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201583 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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d8cc968adc |
Merged revisions 201450 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun 2009) | 9 lines Change the datastore traversal in ast_do_masquerade to use a safe list traversal. It is possible for datastore fixup functions to remove the datastore from the list and free it. In particular, the queue_transfer_fixup in app_queue does this. While I don't yet know of this causing any crashes, it certainly could. Found while discussing a separate issue with Brian Degenhardt. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201458 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4c0265664e |
Merged revisions 200991 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines Improve support for media paths that can generate multiple frames at once. There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201056 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3c23511069 |
Don't claim a char * is a mansession object.
Since there was only 1 bucket, and no hash function was specified, the code actually worked perfectly fine. However, in theory, this was invalid use of the OBJ_POINTER flag, so remove it so the code provides a better usage example. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200805 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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82fb56886e |
More 'static' qualifiers on module global variables.
The 'pglobal' tool is quite handy indeed :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200620 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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6c5987811c |
Redesigned 'optional API' support.
This patch provides a new implementation of the optional API support defined in asterisk/optional_api.h; this new version provides solves compatibility issues with the use of linker version scripts for suppressing global symbols. In addition, there is now a functional (and tested!) implementation for Mac OS/X, so module writers no longer need to use special tests before calling optional API functions. All future implementations must provide these same semantics, so that module writers can rely on them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200519 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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afcbf2e14f |
Merged revisions 200360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun 2009) | 10 lines Suppress a warning message and give a better return code when generating inband ringing after a call is answered. (closes issue #15158) Reported by: madkins Patches: 15158.patch uploaded by mmichelson (license 60) Tested by: madkins ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200361 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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dabfa94fdc |
Release the allocated channel decreasing the reference counter.
When allocating the channel use ao2_ref(-1) to release it, instead of calling ast_free(). Also avoid freeing structures inside that channel (on error) if they will be released by the channel destructor being called if the reference counter reachs 0. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200108 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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df649a8671 |
Fixes the argument order in definition of new_find_extension().
In the definition of new_find_extension(), the arguments 'callerid' and 'label' were swapped. The prototype declaration and all calls to the function are ordered 'callerid' then 'label', but the function itself was ordered 'label' then 'callerid'. (closes issue #15303) Reported by: JimDickenson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199957 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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554456f0fc |
Use ast_channel_unref to instead of ast_free on a newly created channel.
Also I removed an unnecessary free of a cid_name. This will be freed properly in the channel destructor. Reported by mnicholson in #asterisk-dev. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199923 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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d532cbcd8a |
module load priority
This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized. The lower the value, the higher the priority. The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority on load. Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty. (closes issue #15191) Reported by: alecdavis Tested by: dvossel Review: https://reviewboard.asterisk.org/r/262/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199743 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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d1d9beadc9 |
Merged revisions 199297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) | 14 lines Fixes issue with hints giving unexpected results. Hints with two or more devices that include ONHOLD gave unexpected results. (closes issue #15057) Reported by: p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel (license 671) pbx.c.1.4.patch uploaded by p (license 558) devicestate.c.trunk.patch uploaded by p (license 671) Tested by: p_lindheimer, dvossel Review: https://reviewboard.asterisk.org/r/254/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199298 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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befad10893 |
Merged revisions 199022 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines Safely handle AMI connections/reload requests that occur during startup. During asterisk startup, a lock on the list of modules is obtained by the primary thread while each module is initialized. Issue 13778 pointed out a problem with this approach, however. Because the AMI is loaded before other modules, it is possible for a module reload to be issued by a connected client (via Action: Command), causing a deadlock. The resolution for 13778 was to move initialization of the manager to happen after the other modules had already been lodaded. While this fixed this particular issue, it caused a problem for users (like FreePBX) who call AMI scripts via an #exec in a configuration file (See issue 15189). The solution I have come up with is to defer any reload requests that come in until after the server is fully booted. When a call comes in to ast_module_reload (from wherever) before we are fully booted, the request is added to a queue of pending requests. Once we are done booting up, we then execute these deferred requests in turn. Note that I have tried to make this a bit more intelligent in that it will not queue up more than 1 request for the same module to be reloaded, and if a general reload request comes in ('module reload') the queue is flushed and we only issue a single deferred reload for the entire system. As for how this will impact existing installations - Before 13778, a reload issued before module initialization was completed would result in a deadlock. After 13778, you simply couldn't connect to the manager during startup (which causes problems with #exec-that-calls-AMI configuration files). I believe this is a good general purpose solution that won't negatively impact existing installations. (closes issue #15189) (closes issue #13778) Reported by: p_lindheimer Patches: 06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71) Tested by: p_lindheimer, seanbright Review: https://reviewboard.asterisk.org/r/272/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199051 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c42344b319 |
ast_call_forward() todo notes and originate flag copy.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198954 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3830c415c7 |
Generic call forward api, ast_call_forward()
The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options. (closes issue #13630) Reported by: festr Review: https://reviewboard.asterisk.org/r/271/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198856 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4900d37d71 |
Fix a bug where we were passing in address information that should remain unmodified to a function that may modify it.
(closes issue #15243) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198762 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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298d745fb4 |
Add the ability to execute connected line interception macros.
When connected line updates are received or generated in the middle of an application call, it is now possible to execute a macro to manipulate the connected line data. This way, phone numbers may be manipulated to be more presentable to users, names may be changed for...whatever reason, or whatever else needs to be done may be. Review: https://reviewboard.asterisk.org/r/256 AST-165 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198727 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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5d8d1545d8 |
Do not add say.o in a separate line.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198597 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ab73b6e556 |
Fixed an issue in the threadstorage cli functions resulting from the constification of struct ast_cli_args in r196072.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198558 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7ef2d7bca7 |
Filter the say.o object, it is being added later.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198442 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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1630c85861 |
Constification and remove some unused code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198438 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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8580871fd4 |
Constify the ast_frame arg to ast_queue_frame().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198434 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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43dcf1001b |
Simplify the Makefile and avoid needing to specify each object file.
Instead of specifying every object file, use make's magic to generate it. This will generate less conflicts in team branches when a new file is added in trunk. (closes issue #15226) Reported by: eliel Patches: makefile uploaded by eliel (license 64) Review: http://reviewboard.asterisk.org/r/269/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198139 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c8b0c41ed8 |
Merged revisions 198068 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition. This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels. (closes issue #12946) Reported by: meral Patches: null-cdr2.diff uploaded by mnicholson (license 96) Tested by: mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested by: sum ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198072 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c35e305c82 |
Fix a memory leak of the write buffer when writing a file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198064 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f51bb019bb |
Update references to bugs.digium.com and reviewboard.digium.com to the new URLs.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197824 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e4dafd3d0b |
Ensure that accidental calls to ast_string_field_free_memory() on embedded stringfield pools are safe.
It is possible for a stringfield manager structure (and pool) structure to be allocated as part of a larger structure allocation (using ast_calloc_with_strinfields()); when this is done, the stringfield pool cannot be separately freed, but users of the tructure may not be aware (and shouldn't have to be aware) of whether the pool was embedded. This patch modifies the behavior so that they can always call ast_string_field_free_memory() and the function will do the right thing for both embedded and non-embedded situations. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197775 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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98ce0e5b6c |
Eliminate several needless checks and fix a few memory leaks
(closes issue #14833) Reported by: contactmayankjain Patches: all_changes.patch uploaded by contactmayankjain (license 740) slightly modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197616 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ea89db8ce0 |
Fix an incorrect call to ast_string_field_free_memory which caused a crash in the logger.
Since the message structure is allocated using ast_calloc_with_stringfields we do not need to free the memory used for the stringfields as it will get freed when the message structure is. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197570 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |