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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r18087 | kpfleming | 2006-04-06 17:04:58 -0500 (Thu, 06 Apr 2006) | 3 lines
don't try to support 'i' or 'r' options if chan_zap is not loaded, and warn the user when they attempt to use them (issue #6675)
update application help text to more clearly define when Zaptel and chan_zap are required
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application wouldn't block indefinitely looking for another frame from that channel. Don't try to do frame size analysis on a frame that isn't voice, only report DEBUG and VERBOSE msgs to the logger channels when the DEBUG and VERBOSE settings are high enough to require it, and some other minor cleanups.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r17905 | file | 2006-04-06 14:00:10 -0300 (Thu, 06 Apr 2006) | 2 lines
Update email address to my Digium one
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(STATIC_MODULE) to show what needs to be changed.
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rand() to threadsafe ast_random()
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returns differently
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- move "res=-1" out of verbose block.
- minor code cleanup
Imported from 1.2 branch
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https://svn.digium.com/svn/asterisk/branches/1.2
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r17283 | mogorman | 2006-04-04 10:24:50 -0500 (Tue, 04 Apr 2006) | 3 lines
Adds documentation to show what the w flag.
Patch from Ian Kinner at Digium.
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in pbx_exec is always 1 so it can be removed.
This change also takes away ast_exec_extension(), and lets all
switch functions (exists, canmatch, exec, matchmore) all use the same
prototype, which makes the code a bit cleaner.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@16558 f38db490-d61c-443f-a65b-d21fe96a405b
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Thanks everyone involved for hard work, testing and testing!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@16473 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r16192 | tilghman | 2006-03-29 13:11:18 -0600 (Wed, 29 Mar 2006) | 2 lines
Bug 6830 - Let GosubIf work with the same conditions as a GotoIf (change in API approved by Russell)
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On passing, replace strdup with ast_strdup() and remove
the now useless checks for NULL since ast_strdup() can handle
it correctly.
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so that new apps will not try to use it by mistake.
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to S_OR functions. from bug note 6805 with minor
modifications.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@15067 f38db490-d61c-443f-a65b-d21fe96a405b
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various code cleanups
don't steal copyright from author
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on reload.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r14830 | tilghman | 2006-03-24 23:01:16 -0600 (Fri, 24 Mar 2006) | 2 lines
Fix incorrect size of zeroing (left over from when maxmsg was hardcoded at 100)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r14821 | tilghman | 2006-03-24 22:09:04 -0600 (Fri, 24 Mar 2006) | 2 lines
Bug 6783 - When context is specified, voicemail should look for mailboxes in that context
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#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
to these files.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r14659 | bweschke | 2006-03-23 22:53:42 -0600 (Thu, 23 Mar 2006) | 3 lines
Fix a typo in the app description
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r14467 | bweschke | 2006-03-23 14:43:05 -0600 (Thu, 23 Mar 2006) | 3 lines
Bug #5884 - fix a possible race state in app_meetme when a channel has gone away and we are reading continuously for more frames. (mneuhauser)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r14462 | russell | 2006-03-23 15:13:48 -0500 (Thu, 23 Mar 2006) | 2 lines
don't crash when asked to read from a file that doesn't exist (issue #6786)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@14463 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r14276 | file | 2006-03-22 18:18:32 -0400 (Wed, 22 Mar 2006) | 2 lines
Fix a minor code issue
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@14279 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r14275 | file | 2006-03-22 18:13:30 -0400 (Wed, 22 Mar 2006) | 2 lines
Issue #6781 - Verbose levels not enforced in app_voicemail (Reported by flobi)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r13851 | russell | 2006-03-21 10:53:27 -0500 (Tue, 21 Mar 2006) | 2 lines
don't add conference participant if the user hangs up while recording their name (issue #6661)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r13748 | tilghman | 2006-03-21 00:24:56 -0600 (Tue, 21 Mar 2006) | 2 lines
Bug 6714 - Workaround to avoid retrieving incomplete voicemail message
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r13550 | russell | 2006-03-19 04:59:55 -0500 (Sun, 19 Mar 2006) | 4 lines
revert the change made in revision 12927 in favor of keeping the original
behavior of the option. The documentation has now been updated to reflect
the actual behavior. (issue #6523)
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#6722)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r12927 | russell | 2006-03-14 13:41:05 -0500 (Tue, 14 Mar 2006) | 3 lines
when using the G() option to Dial, fix sending the called channel to 1 priority
beyond what was specified (issue #6523)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r12925 | russell | 2006-03-14 13:28:39 -0500 (Tue, 14 Mar 2006) | 3 lines
fix a problem with not loading realtime queue members by always reloading a
realtime queue from the database even if it is found in the list (issue #6680)
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a bug report...)
- Remove unused variable "ret"
- Declare char* pointers in the block where they are used
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r11778 | bweschke | 2006-03-04 06:45:37 -0500 (Sat, 04 Mar 2006) | 3 lines
Substitute variables in the post_process string (if it exists) before those variables could possibly disappear (channel hangup) #6462
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