asterisk

Commit Graph

Author	SHA1	Message	Date
Mark Michelson	630bf109bb	Merged revisions 186832 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186832 \| mmichelson \| 2009-04-07 18:49:49 -0500 (Tue, 07 Apr 2009) \| 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a p2p bridge returns AST_BRIDGE_RETRY. Without this flag set, warning sounds will not be properly played to either party of the bridge. (closes issue #14845) Reported by: adomjan ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	6f53ed4c67	This commit introduces COLP/CONP and Redirecting party information into Asterisk. The channel drivers which have been most heavily tested with these enhancements are chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be introduced in a later commit. chan_skinny has code added to it here, but according to user pj, the support on chan_skinny is not working as of now. This will be fixed in a later commit. A special thanks goes out to bugtracker user gareth for getting the ball rolling and providing the initial support for this work. Without his initial work on this, this would not have been nearly as painless as it was. This functionality has been tested by Digium's product quality department, as well as a customer site running thousands of calls every day. In addition, many many many many bugtracker users have tested this, too. (closes issue #8824) Reported by: gareth Review: http://reviewboard.digium.com/r/201 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	2d9c6ef3d5	Add better support for relaying success or failure of the ast_transfer() API call. This API call now waits for a special frame from the underlying channel driver to indicate success or failure. This allows the return value to truly convey whether the transfer worked or not. In the case of the Transfer() dialplan application this means the value of the TRANSFERSTATUS dialplan variable is actually true. (closes issue #12713) Reported by: davidw Tested by: file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	083e57a5e5	Merged revisions 185771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185771 \| russell \| 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009) \| 6 lines Fix a case where DTMF could bypass audiohooks. This change fixes a situation where an audiohook that wants DTMF would not actually get it. This is in the code path where we end DTMF digit length emulation while handling a NULL frame. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185772 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	9381bff79d	Improve timing interface to remember which provider provided a timer The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error. This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider. (closes issue #14697) Reported by: moy Review: http://reviewboard.digium.com/r/211/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	2ad737608c	Put siren7 and siren14 in ast_best_codec() just so they're in there somewhere. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	10b7b842dc	Fix an issue where a T38 control frame would get dropped. If two channels were bridged together using a generic bridge the T38 control frame would get passed up instead of being indicated on the other channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0bdd99ad64	Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 \| russell \| 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) \| 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182847 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	9d6ba51d05	Tweak the handling of the frame list inside of ast_answer(). This does not change any behavior, but moves the frames from the local frame list back to the channel read queue using an O(n) algorithm instead of O(n^2). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	16b9280ba9	correct logic flaw in ast_answer() changes in r182525 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	d11b6386a5	Improve behavior of ast_answer() to not lose incoming frames ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations. When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames. This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller. http://reviewboard.digium.com/r/196/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182525 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	5308112806	Fix a memory leak in the ast_answer / __ast_answer API call. For a channel that is not yet answered this API call will wait until a voice frame is received on the channel before returning. It does this by waiting for frames on the channel and reading them in. The frames read in were not freed when they should have been. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c61a3f2878	Make handling of the BRIDGE_PLAY_SOUND variable thread-safe. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ffc7510e7a	Make handling of the BRIDGEPVTCALLID variable thread-safe. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	29cfabf335	Merged revisions 181423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181423 \| russell \| 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) \| 9 lines Make code that updates BRIDGEPEER variable thread-safe. It is not safe to read the name field of an ast_channel without the channel locked. This patch fixes some places in channel.c where this was being done, and lead to crashes related to masquerades. (closes issue #14623) Reported by: guillecabeza ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	979eb709ae	app_read does not break from prompt loop with user terminated empty string In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata(). (closes issue #14279) Reported by: Marquis Patches: fix_app_read.patch uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	cfa0d9c0ce	Merged revisions 179741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179741 \| russell \| 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) \| 6 lines Ensure chan->fdno always gets reset to -1 after handling a channel fd event. Since setting fdno to -1 had to be moved, a couple of other code paths that do process an fd event return early and do not pass through the code path where it was moved to. So, set it to -1 in a few other places, too. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179742 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	a65727949c	Merged revisions 179671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179671 \| file \| 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) \| 3 lines Move where fdno is set to the default value to after the read callback of the channel driver is called. We have to do this as the underlying channel driver may need the fdno value to determine what to read. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	d9b034a430	Merged revisions 179608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179608 \| russell \| 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) \| 9 lines Make it easier to detect an improper call to ast_read(). When you call ast_waitfor() on a channel, the index into the channel fds array that holds the file descriptor that poll() determines has input available is stored in fdno. This patch clears out this value after a call to ast_read() and also reports errors if ast_read() is called without an fdno set. From a discussion on the asterisk-dev list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179609 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	aa81288bab	Merged revisions 179536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179536 \| jpeeler \| 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) \| 15 lines Fix bridging regression from commit 176701 This fixes a bad regression where the bridge would exit after an attended transfer was made. The problem was due to nexteventts getting set after the masquerade which caused the bridge to return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: tim_ringenbach ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0c0479602e	Merged revisions 179461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179461 \| russell \| 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) \| 8 lines Ensure that only one thread is calling ast_settimeout() on a channel at a time. For example, with an IAX2 channel, you can have both the channel thread and the chan_iax2 processing threads calling this function, and doing so twice at the same time is a bad thing. (Found in a debugging session with dvossel and mmichelson) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	f40edf2793	Merged revisions 176701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176701 \| jpeeler \| 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) \| 17 lines Modify bridging to properly evaluate DTMF after first warning is played The main problem is currently if the Dial flag L is used with a warning sound, DTMF is not evaluated after the first warning sound. To fix this, a flag has been added in ast_generic_bridge for playing the warning which ensures that if a scheduled warning is missed, multiple warrnings are not played back (due to a feature evaluation or waiting for digits). ast_channel_bridge was modified to store the nexteventts in the ast_bridge_config structure as that information was lost every time ast_channel_bridge was reentered, causing a hangup due to incorrect time calculations. (closes issue #14315) Reported by: tim_ringenbach Reviewed on reviewboard: http://reviewboard.digium.com/r/163/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176708 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c461d29b0b	Update the timing API to have better support for multiple timing interfaces. 1) Add module use count handling so that timing modules can be unloaded. 2) Implement unload_module() functions for the timing interface modules. 3) Allow multiple timing modules to be loaded, and use the one with the highest priority value. 4) Report which timing module is being use in the "timing test" CLI command. (closes issue #14489) Reported by: russell Review: http://reviewboard.digium.com/r/162/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	4ec301360c	Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	96326f5aa1	Make the causes array static, and remove the type name as it is not needed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175983 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	47ebea6a8d	Fix 'd' option for app_dial and add new option to Answer application The 'd' option would not work for channel types which use RTP to transport DTMF digits. The only way to allow for this to work was to answer the channel if we saw that this option was enabled. I realized that this may cause issues with CDRs, specifically with giving false dispositions and answer times. I therefore modified ast_answer to take another parameter which would tell if the CDR should be marked answered. I also extended this to the Answer application so that the channel may be answered but not CDRified if desired. I also modified app_dictate and app_waitforsilence to only answer the channel if it is not already up, to help not allow for faulty CDR answer times. All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the changes except for the change to the Answer application will go in since we do not introduce new features into stable branches (closes issue #14164) Reported by: DennisD Patches: 14164.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/145 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	8435535300	Tell the device state core a change happened when a channel is freed but not a specific state. We need to do this because while we know that the freeing of the channel may cause something to become not in use we do not know this for sure. There may be another channel that is still up which would cause it to be in use. (closes issue #13238) Reported by: kowalma Patches: 20090121__bug13238.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	84053c05c7	Add extensions and context on manager event when new channel is created. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	3fd61d729c	Merged revisions 170648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170648 \| file \| 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) \| 4 lines When a channel is answered make sure any indications currently playing stop. Usually the phone would do this but if the channel was already answered then they are being generated by Asterisk and we darn well need to stop them. (closes issue #14249) Reported by: RadicAlish ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	dccc06063f	Merged revisions 170392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170392 \| mmichelson \| 2009-01-23 09:40:39 -0600 (Fri, 23 Jan 2009) \| 28 lines Fix broken call pickup There was a subtle change in ast_do_masquerade which resulted in failed attempts to pickup calls. The problem was that the value of the AST_FLAG_OUTGOING flag was copied from the clone to the original channel. In the case of call pickup, this meant that the AST_FLAG_OUTGOING flag ended up being cleared on the channel that was attempting to execute the pickup. Because this flag was not set, when ast_read came across an answer frame, it ignored it. The result of this was that the calling channel was never properly answered. This fix changes the behavior in ast_do_masquerade to set the flags on the original channel to the union of the flags on the clone channel. This way, if the AST_FLAG_OUTGOING flag is set on either of the two channels involved in the masquerade, the resulting channel will have the flag set as well. (closes issue #14206) Reported by: francesco_r Patches: 14206.patch uploaded by putnopvut (license 60) Tested by: francesco_r, aragon, putnopvut ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ef6ad2b53c	Merged revisions 168561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 \| russell \| 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) \| 2 lines Revert unnecessary indications API change from rev 122314 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	859ae78977	Merged revisions 166568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166568 \| mmichelson \| 2008-12-23 09:16:26 -0600 (Tue, 23 Dec 2008) \| 12 lines Fix a crash resulting from a datastore with inheritance but no duplicate callback The fix for this is to simply set the newly created datastore's data pointer to NULL if it is inherited but has no duplicate callback. (closes issue #14113) Reported by: francesco_r Patches: 14113.patch uploaded by putnopvut (license 60) Tested by: francesco_r ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	18e07935ed	Merged revisions 166509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166509 \| tilghman \| 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008) \| 4 lines Use the integer form of condition for integer comparisons. (closes issue #14127) Reported by: andrew ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	9733b30ff0	Adding a new dialplan function AUDIOHOOK_INHERIT This function is being added as a method to allow for an audiohook to move to a new channel during a channel masquerade. The most obvious use for such a facility is for MixMonitor when a transfer is performed. Prior to the addition of this functionality, if a channel running MixMonitor was transferred by another party, then the recording would stop once the transfer had completed. By using AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the call even after the transfer has completed. It has also been determined that since this is seen by most as a bug fix and is not an invasive change, this functionality will also be backported to 1.4 and merged into the 1.6.0 branches, even though they are feature-frozen. (closes issue #13538) Reported by: mbit Patches: 13538.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/102/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c9eb01c899	Merged revisions 164201 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164201 \| russell \| 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008) \| 31 lines Handle a case where a call can be bridged to a channel that is still ringing. The issue that was reported was about a case where a RINGING channel got redirected to an extension to pick up a call from parking. Once the parked call got taken out of parking, it heard silence until the other side answered. Ideally, the caller that was parked would get a ringing indication. This patch fixes this case so that the caller receives ringback once it comes out of parking until the other side answers. The fixes are: - Make sure we remember that a channel was an outgoing channel when doing a masquerade. This prevents an erroneous ast_answer() call on the channel, which causes a bogus 200 OK to be sent in the case of SIP. - Add some additional comments to explain related parts of code. - Update the handling of the ast_channel visible_indication field. Storing values that are not stateful is pointless. Control frames that are events or commands should be ignored. - When a bridge first starts, check to see if the peer channel needs to be given ringing indication because the calling side is still ringing. - Rework ast_indicate_data() a bit for the sake of readability. (closes issue #13747) Reported by: davidw Tested by: russell Review: http://reviewboard.digium.com/r/90/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164203 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	035a7552d6	Since chan_sip is callback devicestate driven do not pass in actual states, pass in unknown so we get asked. Additionally do not pass in an actual device state value in ast_setstate since the channel may be callback driven. (closes issue #13525) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	7fcac067b2	Merged revisions 163448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163448 \| russell \| 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) \| 26 lines Resolve issues that could cause DTMF to be processed out of order. These changes come from team/russell/issue_12658 1) Change autoservice to put digits on the head of the channel's frame readq instead of the tail. If there were frames on the readq that autoservice had not yet read, the previous code would have resulted in out of order processing. This required a new API call to queue a frame to the head of the queue instead of the tail. 2) Change up the processing of DTMF in ast_read(). Some of the problems were the result of having two sources of pending DTMF frames. There was the dtmfq and the more generic readq. Both were used for pending DTMF in various scenarios. Simplifying things to only use the frame readq avoids some of the problems. 3) Fix a bug where a DTMF END frame could get passed through when it shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation, and a digit arrived before emulation was complete, digits would get processed out of order. (closes issue #12658) Reported by: dimas Tested by: russell, file Review: http://reviewboard.digium.com/r/85/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	fb242bc8fd	Fix the "failed" extension for outgoing calls. The conversion to use ast_check_hangup() everywhere instead of checking the softhangup flag directly introduced this problem. The issue is that ast_check_hangup() checked for tech_pvt to be NULL. Unfortunately, this will be NULL is some valid circumstances, such as with a dummy channel. The fix is simple. Don't check tech_pvt. It's pointless, because the code path that sets this to NULL is when the channel hangup callback gets called. This happens inside of ast_hangup(), which is the same function responsible for freeing the channel. Any code calling ast_check_hangup() better not be calling it after that point, and if so, we have a bigger problem at hand. (closes issue #14035) Reported by: erogoza git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	48522988ab	In order to move away from nested function use, some changes to the recently introduced ast_channel_search_locked need to be made. Specifically, the caller needs to be able to pass arbitrary data which in turn is passed to the callback. This patch addresses all of the nested functions currently in asterisk trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155590 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	f7c20e0dec	Merged revisions 154685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154685 \| murf \| 2008-11-05 09:06:53 -0700 (Wed, 05 Nov 2008) \| 1 line This fix was prompted by communication from user, who was seeing thousands of error logs... looks like EAGAIN. Made such uninteresting. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	086a52d9d1	Introduce a new API call ast_channel_search_locked, which iterates through the channel list calling a caller-defined callback. The callback returns non-zero if a match is found. This should speed up some of the code that I committed earlier today in chan_sip (which is also updated by this commit). Reviewed by russellb and kpfleming via ReviewBoard: http://reviewboard.digium.com/r/28/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154429 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	bd4eb070f3	bring over all the fixes for the warnings found by gcc 4.3.x from the 1.4 branch, and add the ones needed for all the new code here too git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153616 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	511ce6b2bf	Use the ast_str API call to reset the string instead of manually editing its internals (closes issue #13816) Reported by: eliel Patches: channel.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	fbe13cfb86	Modify ast_answer() to not hold the channel lock while calling ast_safe_sleep() or when calling ast_waitfor(). These are inappropriate times to hold the channel lock. This is what has caused "could not get the channel lock" messages from chan_sip and has likely caused a negative impact on performance results of SIP in Asterisk 1.6. Thanks to file for pointing out this section of code. (closes issue #13287) (closes issue #13115) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	0f0c10993c	Merged revisions 141156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141156 \| murf \| 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) \| 1 line A small change to prevent double-posting of CDR's; thanks to Daniel Ferrer for bringing it to our attention ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	2fceed7f6d	Merged revisions 140690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140690 \| murf \| 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) \| 1 line After reconsidering, with respect to 13409, ast_cdr_detach should be OK, better in fact, than ast_cdr_free, which generates lots of useless warnings that will undoubtably generate complaints. Hmmm. It doesn't hush the useless warnings, but it does allow control of posting via the detach and post routines, for those possible situations, where you'd want to post single-channel cdrs. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	1c79a23b8e	Merged revisions 140670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140670 \| murf \| 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) \| 14 lines (closes issue #13409) Reported by: tomaso Patches: asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license 564) I basically spent the day, verifying that this patch solves the problem, and doesn't hurt in non-problem cases. Why valgrind did not plainly reveal this leak absolutely mystifies and stuns me. Many, many thanks to tomaso for finding and providing the fix. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	fdd92290af	Convert deprecated routines to the new names. (closes issue #13297) Reported by: snuffy Patches: bug13297_20080814.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137456 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	790fde68d9	Another batch of files from RSW. The remaining apps and a few more files from main/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	b69c8e6ab5	Another big chunk of changes from the RSW branch. Bunch of stuff from main/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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280 Commits (630bf109bb3a924e90804cce697b29f2a90d9bd8)