asterisk

Commit Graph

Author	SHA1	Message	Date
Jeff Peeler	3e98181722	Merged revisions 121163 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@121164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	567cb9a041	Merged revisions 114892 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@114893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	d4f34710bf	Merged revisions 114888 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@114889 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	90046983f7	Merged revisions 106239 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@106318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	8f19d8523b	Remove compiler warning for uninitialized variable git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	3a95a07e69	Add a "console active" CLI command, which lets you find out which console device 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: https://origsvn.digium.com/svn/asterisk/trunk@99248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	f20450ea03	Merge changes from team/russell/console_devices - 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: https://origsvn.digium.com/svn/asterisk/trunk@99227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	89778a05c1	Make the output of "console list devices" a bit prettier. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	30de4fc350	List which devices are inputs and outputs in "console list devices" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	2905e69622	Change reference to external library so it appears on the extref listing http://www.asterisk.org/doxygen/trunk/extref.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	f5580269e1	Add the URL to the home page for portaudio. Also add the location of the svn repository to check out portaudio v19. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	fa39abb50b	Add support for generating a ringing sound on an incoming call. This is a bit 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: https://origsvn.digium.com/svn/asterisk/trunk@96079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	b8caf43b3a	Note that this module doesn't actually play a ringing sound for an incoming call ... oops git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	d6c016a5aa	Show the correct CLI command to answer the call git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96076 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	067fcc2a03	Update chan_console to natively use a 16 kHz sample rate. If it is talking 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: https://origsvn.digium.com/svn/asterisk/trunk@95527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	4e99cc88e2	Merge the main set of changes from team/russell/chan_console. 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: https://origsvn.digium.com/svn/asterisk/trunk@95412 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

16 Commits (feead8b7574cc8da437d44f709dff8cd5e885090)