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${ noResults }
3096 Commits (115937b7a6a2bc5d88c39836e98ffed073204a08)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Mark Michelson
|
ed6323cb73 |
Merged revisions 133169 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) | 12 lines As suggested by seanbright, the PSEUDO_CHAN_LEN in app_chanspy should be set at load time, not at compile time, since dahdi_chan_name is determined at load time. Also changed the next_unique_id_to_use to have the static qualifier. Also added the dahdi_chan_name_len variable so that strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for the suggestion. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133171 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
cca455b0f3 |
Merged revisions 133104 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133104 | mmichelson | 2008-07-23 14:06:16 -0500 (Wed, 23 Jul 2008) | 5 lines Zap/pseudo is ten characters, but DAHDI/pseudo is twelve. The strncmp call in next_channel should account for this. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133106 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
1908413bd6 |
Merged revisions 133101 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133101 | mmichelson | 2008-07-23 13:57:17 -0500 (Wed, 23 Jul 2008) | 6 lines Update the "last" channel in next_channel in app_chanspy so that the same pseudo channel isn't constantly returned. related to issue #13124 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133102 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Brett Bryant
|
16374ad40d |
Fix a bug where SENDTEXTSTATUS isn't set properly when it isn't
supported on a channel (yet _another_ useful patch by eliel). (closes issue #13081) Reported by: eliel Patches: app_sendtext.c.patch uploaded by eliel (license 64) Tested by: eliel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132508 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Russell Bryant
|
c87f901cfd |
Remove libresample from the Asterisk source tree. It is now available in its
own repository, and must be installed like any other library for Asterisk to use. The two modules that require it are codec_resample and app_jack. To install libresample: $ svn co http://svn.digium.com/svn/libresample/trunk libresample $ cd libresample $ ./configure $ make $ sudo make install This code is currently in our own repository because the build system did not include the appropriate targets for building a dynamic library or for installing the library. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132390 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
|
5de127e103 |
Enable higher quality resampling, as it doesn't have a noticeable performance
impact on my machine .. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132388 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
84282953d8 |
Document that the duration of dtmf may be passed to
the SendDTMF application. Also correct the default pause between digits. (closes issue #13102) Reported by: eliel Patches: app_senddtmf.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131824 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Brett Bryant
|
8d95aefbde |
Janitor project: convert free to ast_free
(closes issue #13082) Reported by: eliel Patches: app_rpt.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131529 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
|
b4dac0b385 |
Merged revisions 131369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131369 | mmichelson | 2008-07-16 15:23:02 -0500 (Wed, 16 Jul 2008) | 14 lines Move the init_queue call back to where it used to be (changed Sept 12 last year). It was moved then to prevent a memory leak. Since then, the same memory leak recurred and was fixed in a better way. Now it has been found that the placement of this init_queue call can cause problems if a realtime queue has values changed to an empty string. The problem is that the default value for that queue parameter would not be set. (closes issue #13084) Reported by: elbriga ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131375 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
|
b35c06d838 |
Merged revisions 131357 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131357 | mmichelson | 2008-07-16 14:37:08 -0500 (Wed, 16 Jul 2008) | 6 lines Apparently, "thread safety" is important, whatever that means. :P (Thanks Russell!) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131358 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
|
b95bc53c23 |
Merged revisions 131299 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131299 | mmichelson | 2008-07-16 13:57:34 -0500 (Wed, 16 Jul 2008) | 13 lines Make absolutely certain that the transfer datastore is removed from the calling channel once the caller is finished in the queue. This could have weird con- sequences when dialing local queue members when multiple transfers occur on a single call. Also fixed a memory leak that would occur when an attended transfer occurred from a queue member. (closes issue #13047) Reported by: festr ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131300 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Tilghman Lesher
|
49715c05f1 |
Merged revisions 130959 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) | 8 lines astman_send_error does not need a newline appended -- the API takes care of that for us. (closes issue #13068) Reported by: gknispel_proformatique Patches: asterisk_1_4_astman_send.patch uploaded by gknispel (license 261) asterisk_trunk_astman_send.patch uploaded by gknispel (license 261) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131044 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Brett Bryant
|
fe874bfe6b |
Fix memory leak in app_queue when a device state is changed but it isn't
a member of any queue. (closes issue #13073) Reported by: eliel Patches: app_queue.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131015 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
|
bd1bb0d0e2 |
Merged revisions 130792 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul 2008) | 8 lines Add a check to the CAN_EARLY_BRIDGE macro in app_dial to be sure there are no audiohooks present on the channels involved. This fixed a one-way audio situation I had in my test setup. I couldn't find any open issues that suggested one-way audio with regards to mixmonitor (or other audiohook) usage, though. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130794 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Brett Bryant
|
5b7933fe5e |
Janitor patch to change uses of sizeof to ARRAY_LEN
(closes issue #13054) Reported by: pabelanger Patches: ARRAY_LEN.patch2 uploaded by pabelanger (license 224) Tested by: seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130129 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
|
1c49415b19 |
Removed the fn2 field from the vm_state structure.
fn2 was used in three functions. In every case, it was initialized in the function it was used in. This meant there was no need to have it in a malloc'd structure just taking up space. Furthermore two of the functions it was used in were completely unnecessary since fn2 was set to exactly the same value as the vm_state's fn string. fn2 was a char array sized at PATH_MAX. On my system, PATH_MAX is 4096. This equates to a 4K memory savings per vm_state allocated. Since there is a vm_state malloc'd for every voicemail user on the system, this could potentially add up nicely if there are lots of users. In addition, a vm_state is allocated on the stack each time a caller calls the VoiceMailMain application, meaning that there is a significant stack savings with this patch too. Of course, a single vm_state struct still takes up approximately 20K on my system (when using IMAP storage. Without IMAP storage, there would be about another 300 bytes fewer usage), even with this removal. Further optimizations are probably possible, but most likely not as easy as this one. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129734 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Brett Bryant
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7588bcf690 |
Fixes a bug where the interface for a queue member gets reloaded as the state_interface, if a state_interface was set, on reload because the
state_interface isn't stored in the ast_db. (closes issue #13043) Reported by: jvandal Patches: app_queue.patch uploaded by jvandal (license 413) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129684 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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6ddcd21a7f |
Fix compilation error when IMAP storage is enabled
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129270 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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da03cdd174 |
Merged revisions 129149 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) | 8 lines Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not registered. (closes issue #12885) Reported by: ibc Patches: 20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14) Tested by: ibc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129152 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Brett Bryant
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d185405755 |
Janitor project to convert sizeof to ARRAY_LEN macro.
(closes issue #13002) Reported by: caio1982 Patches: janitor_arraylen5.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129045 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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2ed6cea551 |
Update app_fax for better compatibility with spandsp 0.0.5. Add a call to
t38_terminal_release, and make sure that the phase E handler gets called with proper status. (closes issue #13020) Reported by: dimas Patches: v1-appfax.patch uploaded by dimas (license 88) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129006 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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8fa66db120 |
Merged revisions 128856 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128856 | tilghman | 2008-07-07 19:01:30 -0500 (Mon, 07 Jul 2008) | 7 lines Check for non-NULL before stripping characters. (closes issue #12954) Reported by: bfsworks Patches: 20080701__bug12954.diff.txt uploaded by Corydon76 (license 14) Tested by: deti ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128857 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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5a71f180ad |
Merged revisions 128812 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128812 | tilghman | 2008-07-07 18:21:52 -0500 (Mon, 07 Jul 2008) | 2 lines Stop using deprecated method, as requested by Kevin. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128830 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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4f0f9d27cf |
Crap
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128733 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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742815875d |
If imapfolder=foo were set in voicemail.conf, then when calling VoiceMailMain,
app_voicemail would attempt to play a file called vm-foo instead of playing vm-INBOX to play the "new" sound file. This commit fixes that issue. This may fix one of the problems reported in issue #12987 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128731 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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04a10e77a0 |
Get app_voicemail compiling when IMAP storage is used.
Brought up by reporter on issue #12987 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128690 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Kevin P. Fleming
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6b06e9a8eb |
Merged revisions 127892,127895 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127892 | kpfleming | 2008-07-03 17:18:38 -0500 (Thu, 03 Jul 2008) | 6 lines a couple of small Solaris-related fixes (closes issue #11885) Reported by: snuffy, asgaroth ........ r127895 | kpfleming | 2008-07-03 17:20:16 -0500 (Thu, 03 Jul 2008) | 3 lines remove this, it has been moved to the main Makefile ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127903 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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6cfe27089c |
Make change proposed by andrew53 on bugtracker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127857 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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5a8a23f9ba |
Thanks to a suggestion from seanbright, print a warning if the attachment
of the whisper or barge audiohooks fails. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127856 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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bb9a355ff0 |
Fix build
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127852 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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dd315e5f74 |
Fix a crash when attempting to spy on an unbridged channel.
(closes issue #12986) Reported by: andrew53 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127831 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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e4c93fc8c3 |
Added a new option, "timeoutpriority" to queues.conf. A detailed
explanation of the change may be found in configs/queues.conf.sample (closes issue #12690) Reported by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127720 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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267b9d4eb4 |
Oops
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127622 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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885d17506b |
Keep ast_app_inboxcount API compatible with 1.6.0.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127609 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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3a27a6a9e7 |
Fix some crashlike bugs because flag could be NULL in play_record_review().
(Closes issue #12892) Reported by: jaroth Patch originally by jaroth, fixed by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127564 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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13a81dd7ae |
Merged revisions 127244 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127244 | mmichelson | 2008-07-01 18:36:40 -0500 (Tue, 01 Jul 2008) | 5 lines Add error message to failed open(2) calls inside the copy() function of app_voicemail. This idea came as part of my work in helping to resolve issue #12764. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127245 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
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da14954bdc |
another minor ast_channel memory size decrease... for nearly all channels, 'dialcontext' is only going to be set once during the channel's lifetime, so make it a string field instead of a char array
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126960 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Luigi Rizzo
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f55143f0a6 |
fix an uninitialized variable
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126571 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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6d1ebfbed5 |
Remove debug message
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125856 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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ab7809ace9 |
Ensure the thread-safety of the monexec variable in app_queue.
Thanks to Russell for pointing out the problem git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125855 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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09c659d7d0 |
Make this compile with dev-mode on
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125666 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
|
9aef593e58 |
The monitor-join option for queues was deprecated in favor of using
MixMonitor to mix audio. However, it was pointed out to me that because of this, the command set for the MONITOR_EXEC variable is ignored as well. This means that people can't do their own custom mixing commands at the end of recordings in order to make, for instance, stereo recordings of calls. With this patch, app_queue will set the "joinfiles" variable for the channel's monitor if MONITOR_EXEC is not zero-length. This means that for normal audio mixing, MixMonitor is still the preferred choice, but we allow custom mixing to be done with the two Monitor streams if desired. (closes issue #12923) Reported by: snyfer git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125649 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
|
0178d0ccd6 |
Improve consistency between app_dial and app_queue with regards
to how language is handled between two channels whose native language is different. Prior to this patch, app_dial would have the callee inherit the caller's language, and app_queue would not. After this patch, app_dial no longer has the language inheritance capability. This seems to make the most sense since it seems more natural for a person to hear files played back in his/her native language instead of the language of the person on the far end of the call. See the CHANGES file for hints on how to keep the previous behavior of app_dial if desired. (closes issue #12489) Reported by: bcnit git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125647 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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edbe6b7a25 |
Fix a really stupid mistake
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125591 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
|
3859074667 |
Merged revisions 125585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125585 | mmichelson | 2008-06-26 17:52:39 -0500 (Thu, 26 Jun 2008) | 11 lines Add the interface of a queue member to the output of the "queue show" command so that it can easily be associated with a queue member's name. This helps so that the appropriate queue member can be removed or paused since the interface is required, not the member's name. (closes issue #12783) Reported by: davevg Patches: app_queue.diff uploaded by davevg (license 209) with small mod from me ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125586 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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96f92f468f |
Merged revisions 125476 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125476 | mmichelson | 2008-06-26 15:56:01 -0500 (Thu, 26 Jun 2008) | 11 lines Prior to this patch, the "queue show" command used cached information for realtime queues instead of giving up-to-date info. Now realtime is queried for the latest and greatest in queue info. (closes issue #12858) Reported by: bcnit Patches: queue_show.patch uploaded by putnopvut (license 60) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125477 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
|
c9ac1b8ca5 |
Don't play "your message has been saved" twice.
(closes issue #12893) Reported by: jaroth Patches: duplicate_saved.patch uploaded by jaroth (license 50) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125438 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
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fd4a60c459 |
Merged revisions 125132 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it get app_rpt building again after the DAHDI changes (closes issue #12911) Reported by: tzafrir ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125138 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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7ec25255c5 |
Merged revisions 124910 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124910 | tilghman | 2008-06-24 16:08:52 -0500 (Tue, 24 Jun 2008) | 8 lines Occasionally control characters find their way into CallerID. These need to be stripped prior to placing CallerID in the headers of an email. (closes issue #12759) Reported by: RobH Patches: 20080602__bug12759__2.diff.txt uploaded by Corydon76 (license 14) Tested by: RobH ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124912 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Sean Bright
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3e0071199d |
Let app_rpt compile.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124596 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |