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r52137 | russell | 2007-01-24 18:39:50 -0600 (Wed, 24 Jan 2007) | 3 lines
Fix a seg fault when running this application with no arguments from AGI.
(issue #8905, junky)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) | 6 lines
Fix an issue related to synchronization of recordings when using Monitor().
The bug is a miscalculation of the amount to seek the stream for writing to
disk when the number of samples coming in and out of a channel do not match up.
(issue #8298, #8887, report and patch by guillecabeza, patch files created and
testing done by whoiswes)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when sending some sort of response, or calling one of the manager action
callbacks. This resolves an issue where people using the GUI would get random
crashes when they start clicking around a lot.
(issue #8711, reported and debugged by zandbelt)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
initialized to the list head *after* locking the list. Also, lock the actions
list in one place it is being accessed where it was not being done.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r50495 | crichter | 2007-01-11 14:27:52 +0100 (Do, 11 Jan 2007) | 6 lines
* more additions to make the RESTART message work
* added fix for misdn_call to allow SETUPs with empty
extensions, replaced the strtok_r functions with strsep for that
(inspired by Sandro Cappellazzo, thanks)
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r50506 | crichter | 2007-01-11 15:45:38 +0100 (Do, 11 Jan 2007) | 1 line
when we get L2 UP, the L1 is UP definitely too, so we set the L1 state up as well.
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r51410 | russell | 2007-01-22 13:39:30 -0600 (Mon, 22 Jan 2007) | 3 lines
Merge codec_zap support for the transcoder card. This is a standalone codec
module so it will not affect anything else.
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r51359 | file | 2007-01-22 11:23:03 -0500 (Mon, 22 Jan 2007) | 2 lines
Explicitly declare what codecs are supported by default globally since using a bitmask for all may include ones we don't need. (issue #8357 reported by gknispel_proformatique)
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"2131" would've translated to something along the lines of (pardon my..Italian {or lack thereof})
"duecentocentotrentuno", which makes no sense at all.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Properly support 21, 31, 41, 51, 61, 71, 81, and 91.
einundzwanzig has the same format as zweiundzwanzig (as do all other "_ZX" spoken numerals)
Fix support for numbers in the 10,000,000 to 99,999,999 range.
Add support for numbers in the 100,000,000 to 999,999,999 range.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
actually the same thing. So, a digit would have been interpreted incorrectly
here. Since the channel driver will always have the begin and end callbacks
called for a digit, only support the button-down and button-up messages.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/branches/1.4@51311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
curses, termcap, or tinfo are further passed along to the editline configure
script. This fixes some cross-compilation environments.
(issue #8637, reported by ovi, patch by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
