asterisk

Commit Graph

Author	SHA1	Message	Date
Jeff Peeler	33b3d38a3c	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/trunk@121163 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	6d018f0774	chan_console fixes because of ast_frame.data => ast_frame.data.ptr git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	f1e9371da8	- revert change to ast_queue_hangup and create ast_queue_hangup_with_cause - make data member of the ast_frame struct a named union instead of a void Recently the ast_queue_hangup function got a new parameter, the hangupcause Feedback came in that this is no good and that instead a new function should be created. This I did. The hangupcause was stored in the seqno member of the ast_frame struct. This is not very elegant, and since there's already a data member that one should be used. Problem is, this member was a void *. Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone wants to store another type in there in the future. This commit is so massive, because all ast_frame.data uses have to be altered to ast_frame.data.data Thanks russellb and kpfleming for the feedback. (closes issue #12674) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	59f170973e	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/trunk@114892 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	7cfd8389ac	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/trunk@114888 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	f98b2cfef9	Speaking of building... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	41fd7a6a21	(closes issue #6113 ) Reported by: oej Tested by: jpeeler This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option. Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	496adc6fc0	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/trunk@106239 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

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70 Commits (aa5707b8891119741baae98cf7b51a00c0f99438)