Olle Johansson [Thu, 1 Feb 2007 21:05:34 +0000 (21:05 +0000)]
- Clean INC_COUNT flag when we decrement call counter
- If it's still set at time of dialog destruction, make sure we decrement the device call counter properly
before we destroy the dialog
Joshua Colp [Thu, 1 Feb 2007 00:24:20 +0000 (00:24 +0000)]
When going on hold have the side that was put on hold reinvite back to Asterisk. When going off hold have the side that was taken off hold reinvited back to the other party.
Russell Bryant [Wed, 31 Jan 2007 17:34:22 +0000 (17:34 +0000)]
Instead of always creating a realtime queue member as unpaused, read the
"paused" column and use that value for the paused status of the member.
(issue #8949, jmls)
Russell Bryant [Tue, 30 Jan 2007 23:23:24 +0000 (23:23 +0000)]
When we are checking for a system installed version of libgsm, we need to check
for gsm.h as well. Furthermore, when checking for this header, it may be
located in a gsm/ sub directory, so check for that, as well.
(issue #8773)
Russell Bryant [Tue, 30 Jan 2007 19:43:59 +0000 (19:43 +0000)]
Blocked revisions 52954 via svnmerge
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r52954 | russell | 2007-01-30 13:41:52 -0600 (Tue, 30 Jan 2007) | 4 lines
Don't print a message indicating that we don't know what to do with a proceeding
control frame in ast_request_and_dial(). We just need to ignore it.
(reported by JerJer on #asterisk-dev)
Russell Bryant [Tue, 30 Jan 2007 19:33:12 +0000 (19:33 +0000)]
Only set the DTMF flag on the rtp structure if the DTMF mode is actually
RFC2833, not just that it is not INFO. This makes it get set for inband DTMF
as well, which is not valid.
(issue #8936)
Russell Bryant [Tue, 30 Jan 2007 17:19:39 +0000 (17:19 +0000)]
Merged revisions 52903 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 Jan 2007) | 9 lines
The SIGHUP handler was implemented to allow admins to send SIGHUP to a running
Asterisk process to reload the configuration. However, doing the actual reload
in the signal handler itself is a very bad thing to do, because the reload
process includes calling non-reentrant functions such as malloc/calloc/etc.
If Asterisk is running in the background, then the reload will happen
immediately. However, if running in console mode, the reload doesn't work
until something is typed at the console. That sort of defeats the purpose,
but I don't see an easy way to get around it at this point.
Russell Bryant [Tue, 30 Jan 2007 15:36:36 +0000 (15:36 +0000)]
Blocked revisions 52857 via svnmerge
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r52857 | russell | 2007-01-30 09:35:23 -0600 (Tue, 30 Jan 2007) | 5 lines
Comment out the parts in the Makefile that make codec_zap get built. It will
not yet build against zaptel 1.2, so I am disabling it to prevent further bug
reports until it gets merged.
(issue #8940)
Russell Bryant [Tue, 30 Jan 2007 00:15:50 +0000 (00:15 +0000)]
Merged revisions 52762 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) | 5 lines
Fix the extraction of the timestamp from video frames. It was using the
mapping for a mini-frame instead of a video-frame, which caused it to
get invalid data.
(issue #8795, mihai)
Now that filename is part of the structure and since it comes before postprocess... we have to add it to our postprocess line. (reported on asterisk-dev by Boris Bakchiev)
Russell Bryant [Mon, 29 Jan 2007 21:26:27 +0000 (21:26 +0000)]
Fix a problem with packet-to-packet bridging and DTMF mode translation. P2P
bridging can only be used when the DTMF modes don't match if the core is
monitoring DTMF in both directions. Then, the core will handle the translation.
Otherwise, this bridging method can not be used.
(issue #8936)
Russell Bryant [Mon, 29 Jan 2007 20:39:20 +0000 (20:39 +0000)]
The session lock can not be held while calling action callbacks. If so, then
when the WaitEvent callback gets called, then no event can happen because the
session can't be locked by another thread. Also, the session needs to be
locked in the HTTP callback when it reads out the output string. This fixes
the deadlock reported in both 8711 and 8934.
Regarding issue 8711, there still may be an issue. If there is a second action
requested before the processing of the first action is finished, there could
still be some corruption of the output string buffer used to build the result.
(issue #8711, #8934)
Steve Murphy [Mon, 29 Jan 2007 17:57:08 +0000 (17:57 +0000)]
this is for 8778 (pt_BR backport to 1.4). It was committed to trunk via 7663. But it wasn't so much an enhancement as a fix for the bad language output for portuguese in Brazil, so, after a lot of prodding from patient Brazilians, here is the same fix for 1.4
Russell Bryant [Mon, 29 Jan 2007 16:54:27 +0000 (16:54 +0000)]
Clean up a few things in the last commit to the adaptive jitterbuffer code.
- Specifically indicate to the compiler that the "dropem" variable only
needs one but.
- Change formatting to conform to coding guidelines.
Jim Dixon [Mon, 29 Jan 2007 04:18:36 +0000 (04:18 +0000)]
Fixed problem with jitterbuf, whereas it would not complain about, and
would allow itself to be overfilled (per the max_jitterbuf parameter). Now
it rejects any data over and above that size, and complains about it.
Joshua Colp [Sat, 27 Jan 2007 00:44:51 +0000 (00:44 +0000)]
Have the manager interface send back an "Already logged in" message instead of "Invalid/Unknown Command" when the client authenticates for a second time. (issue #8509 reported by pari)
Steve Murphy [Wed, 24 Jan 2007 18:26:22 +0000 (18:26 +0000)]
Merged revisions 52002 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1 line
updated check_expr via 8322 (refactoring of expression checking impl); elfring contributed a nice code reorg, I contributed some time to get it working again, better messages
........
Joshua Colp [Wed, 24 Jan 2007 17:59:55 +0000 (17:59 +0000)]
Fix changing channel formats when joint capability changes and there are no audio formats... I didn't break it originally! (issue #8535 reported by ivoc)
Russell Bryant [Wed, 24 Jan 2007 00:59:58 +0000 (00:59 +0000)]
Merged revisions 51843 via svnmerge from
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)
Russell Bryant [Wed, 24 Jan 2007 00:19:55 +0000 (00:19 +0000)]
Merged revisions 51828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23 Jan 2007) | 4 lines
Don't set a new value for the END_ variable on the channel before using the
old value. If you do, it will lead to accessing a memory address that has
been free()'d. (issue #8895, arkadia)
Russell Bryant [Tue, 23 Jan 2007 22:04:01 +0000 (22:04 +0000)]
Fix some bugs in process_message(). The manager session lock needs to be held
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)
Russell Bryant [Tue, 23 Jan 2007 21:33:15 +0000 (21:33 +0000)]
When traversing the list of manager actions, the iterator needs to be
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.
Steve Murphy [Tue, 23 Jan 2007 18:58:27 +0000 (18:58 +0000)]
via 8748 (callerid.c loses name when returning PRIVATE_NUMBER flag), the user suggested this mod, saying it would allow 'WITHHELD' to appear in the name field, which would be useful
* 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)
........
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.
........
Move filestream creation to Mixmonitor loop. This will prevent a blank file from being created if no frames ever pass through to be recorded. (issue #7589 reported by steve_mcneil)
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)
Jason Parker [Sat, 20 Jan 2007 06:53:49 +0000 (06:53 +0000)]
Fix Italian numeral support in say.conf for "_[2-9]00" case.
"2131" would've translated to something along the lines of (pardon my..Italian {or lack thereof})
"duecentocentotrentuno", which makes no sense at all.
Russell Bryant [Fri, 19 Jan 2007 19:08:25 +0000 (19:08 +0000)]
Fix VLDTMF support in chan_gtalk. AST_FRAME_DTMF and AST_FRAME_DTMF_END are
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.
Russell Bryant [Fri, 19 Jan 2007 17:49:38 +0000 (17:49 +0000)]
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.
Russell Bryant [Fri, 19 Jan 2007 16:56:17 +0000 (16:56 +0000)]
Merged revisions 51300 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19 Jan 2007) | 4 lines
Fix a memory leak on command line tab completion. The container for the
matches was freed, but the individual matches themselves were not.
(issue #8851, arkadia)
issue 7877: chan_zap module reload does not use default/initialized values on subsequent loads. Reset configuration variables to default values prior to parsing configuration file.
projects / thirdparty / asterisk.git / log
