Age | Commit message (Collapse) | Author | Files | Lines |
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r135439 | russell | 2008-08-04 10:02:12 -0500 (Mon, 04 Aug 2008) | 4 lines
Be explicit that we don't want a result from this callback. The callback would
never indicate a match, so nothing would have been returned anyway, but it was
still a poor example of proper usage.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@135440 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r121163 | jpeeler | 2008-06-07 20:41:59 -0500 (Sat, 07 Jun 2008) | 4 lines
This was accidentally reverted.
Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@121164 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r114892 | russell | 2008-04-30 11:34:24 -0500 (Wed, 30 Apr 2008) | 36 lines
Merged revisions 114891 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r114891 | russell | 2008-04-30 11:30:01 -0500 (Wed, 30 Apr 2008) | 28 lines
Merge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4
These changes address a critical performance issue introduced in the latest
release. The fix for the latest security issue included a change that made
Asterisk randomly choose call numbers to make them more difficult to guess by
attackers. However, due to some inefficient (this is by far, an understatement)
code, when Asterisk chose high call numbers, chan_iax2 became unusable after
just a small number of calls. On a small embedded platform, it would not be
able to handle a single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't
run more than about 16 IAX2 channels. Ouch.
These changes address some performance issues of the find_callno() function
that have bothered me for a very long time. On every incoming media frame,
it iterated through every possible call number trying to find a matching
active call. This involved a mutex lock and unlock for each call number
checked. So, if the random call number chosen was 20000, then every media
frame would cause 20000 locks and unlocks. Previously, this problem was
not as obvious since Asterisk always chose the lowest call number it could.
A second container for IAX2 pvt structs has been added. It is an astobj2
hash table. When we know the remote side's call number, the pvt goes into
the hash table with a hash value of the remote side's call number. Then,
lookups for incoming media frames are a very fast hash lookup instead of an
absolutely insane array traversal.
In a quick test, I was able to get more than 3600% more IAX2 channels
on my machine with these changes.
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@114893 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r114888 | jpeeler | 2008-04-30 11:14:43 -0500 (Wed, 30 Apr 2008) | 3 lines
Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@114889 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r106239 | file | 2008-03-05 16:43:22 -0600 (Wed, 05 Mar 2008) | 12 lines
Merged revisions 106235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines
Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things.
(closes issue #12148)
Reported by: jcomellas
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@106318 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99384 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
is currently active for the Asterisk CLI, or to set it. Also, knock multiple device
support off of the to-do list.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99248 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
- Add support for multiple devices. All devices are configured in console.conf.
- Add "console list devices" CLI command to show configured devices. Also, changed
the old "list devices" to be "list available", which queries PortAudio for all
audio devices that are available for use.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99227 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99011 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99009 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
http://www.asterisk.org/doxygen/trunk/extref.html
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97200 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
svn repository to check out portaudio v19.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96692 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
of a hack. It just asks the core to generate the same tone that it would when
you hear ringback when making an outbound call. But hey, it works, and you get
the localized ring tone for the appropriate language set on the channel.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96079 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
... oops
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96077 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96076 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
to an 8 kHz endpoint, then codec_resample will automatically be used to properly
resample the audio before sending it to/from chan_console.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95527 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Add a new console channel driver, chan_console, which is a console channel
driver that uses portaudio as a cross platform audio interface. It was written
to provide a console channel driver that works with Mac CoreAudio, but it
supports a number of other audio interfaces, as well, including OSS and ALSA.
It could one day be the single console channel driver, but does not yet have
as many features as chan_oss.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95412 f38db490-d61c-443f-a65b-d21fe96a405b
|